• RELEVANCY SCORE 5.41

    DB:5.41:Route Groups And H.323 fp






    having trouble building route lists/groups with my multiple h.323 gateways. Basically the h.323 gateways do not show up as possible choices in the dropdown for device when I go to add gateways to a newly created route group. I have seen in numerous documents where h.323 gateways are configured with route groups. Something I am missing? Callmanager version is 3.12c

    DB:5.41:Route Groups And H.323 fp


    A gateway can only belong to a single Route Group. So ensue that you haven't already made that gateway a member of a route group.

    Also if there is a route pattern configured pointing directly to a gateway, you will not be able to add that gateway to a route group.

    Hope this helps,

    Tim Medley, CCNP+Voice, CCDA, CWNA

    Sr. Network Architect

    VoIP Group

    iReadyWorld

  • RELEVANCY SCORE 4.50

    DB:4.50:Route Lists And H.323 Gateways dd






    Have a customer that has the following configuration and I want some conformation that it is working the way the customer thinks it is. We did not create this and one thing is that we will be moving them to MGCP in the future.

    Route Pattern points to a Route List

    Route List points to Route Group A and then Route Group B.

    Route Group A has one H.323 Gateway, AA

    Route Group B has two H.323 Gateways, BA BB

    All Gateways are PRI

    Questions:

    1) Will Call Manager ever get notification if there are no available ports in the AA Gateway?

    2) If the above answer is yes, How long will it take for Gateway AA to tell Call Manager?

    Thanks.

    DB:4.50:Route Lists And H.323 Gateways dd


    Please go thro the below link which will help you in configuring the Route Lists

    Understanding Route Plans

    http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/3_1/sys_ad/adm_sys/ccmsys/a03rp.htm#90437

    Route Group Configuration

    http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/3_1/sys_ad/adm_sys/ccmcfg/b03rtgrp.htm

  • RELEVANCY SCORE 3.98

    DB:3.98:Switching To H.323 From Mgcp jx






    Can anyone give me a quick list of pros and cons of switching to H.323 from MGCP. Someone from Cisco recommended that we switch from MGCP to H.323 in order to resolve some of the issues we're seeing on our PSTN gateways (3745s).

    DB:3.98:Switching To H.323 From Mgcp jx


    You are alos loosing single point of management/configuration. With MGCP routing can be configured from CallManager, no need to replicate configutration on the gateway, with H323 you have to worry about dial-peers everytime you add new external route pattern.

    Chris

  • RELEVANCY SCORE 3.82

    DB:3.82:No Of H.323 Gateways Supported By A Ccm 3s



    Hi,

    Can anyone tell me, how many H.323 Gateways (Netmeeting) will a Call Manager can support??

    Thanks in Advance,

    Naveen

    DB:3.82:No Of H.323 Gateways Supported By A Ccm 3s


    The following is a 'cut' from the "Cisco IP Telephony Solution Reference Network Design Guide" from the chapter entitled, "Cluster Operation and Scalability Guidelines". The guide targets CCM 3.1 and 3.2 and is dated May 2002.

    "You may have up to 500 H.323 (gateway and client) and digital MGCP gateway devices per cluster."

  • RELEVANCY SCORE 3.77

    DB:3.77:Looking For Gatekeeper Redundancy Suggestions as



    Given the following topography, what would be the best way to provide call admission control redundancy for H.323 voice gateways? The attached .gif file shows the layout, here are the basic points:

    * Hub and spoke design where international sites are spokes with one H.323 gateway per site

    * Each international location is a zone.

    * Hub is comprised of three symmetrically configured voice gateways (each with one T1 to software defined network, another to local access) .

    * Two centralized gatekeeper(s) in hub site serving all international voice sites.

    * Gateways and gatekeepers are located in different buildings for diversity, on seperate subnets - within Gigabit Ethernet distance of each other.

    * See drawing for more detail.

    What would be the most appropriate redundancy configuration, given this detail?

    Thanks!

    Tim Moffett

    * Centralized

    DB:3.77:Looking For Gatekeeper Redundancy Suggestions as


    Thanks Ketan,

    I have seen that example before. I now believe that it is the only one in existence!

    Tim

  • RELEVANCY SCORE 3.65

    DB:3.65:Mgcp Gateway kz



    Hi,

    I want to connect two CallManagers via a H.323 trunk.

    CM1 - 2821 Gateway A - 2821 Gateway B - CM2

    now, the gateways will be linked by IP on the local LAN. I am setting up MGCP in CallManager to control the gateways, but not having any luck. How can I create and endpoint seeing that the routers are simply plugging into the local LAN?

    Thanks in advance,

    DB:3.65:Mgcp Gateway kz


    MGCP doesnt make any sense here.

    MGCP protocol was not designed for this situations use simple H323 GW or ICT in CCM.

    Normally with CCM you use it for VG connecting to PSTN.

  • RELEVANCY SCORE 3.61

    DB:3.61:Tcs Record From Mcu j3



             Hello

    I have TCS(ip:172.17.2.102) and MCU (ip:172.17.2.101) and not have h.323 gatekeeper.

    When trying to install new:

    -  "Create recording"

         - dial number 172.17.2.101

         - call type - h.323

    have "Call failed: No route to destination"

    DB:3.61:Tcs Record From Mcu j3


    Everything works.

    Since on h.323 that works with default Templates.

  • RELEVANCY SCORE 3.61

    DB:3.61:I Cannot Create Route Group In Ccm 3.3 (4) pc



    I want to create a route group in the CCM 3.3 but the gateways that have been configured doesn't appear (H.323) so they cannt be inserted in the route groups that I want to create.

    I show some screen shoots for further explanation. Please help me

    Thanks folks

    DB:3.61:I Cannot Create Route Group In Ccm 3.3 (4) pc


    your recommendations helped me to solve part of the problem.

    I am sending a new case, I wait you help me to complete the scenario

    thank you

  • RELEVANCY SCORE 3.52

    DB:3.52:H.323 Gateway And V.150 zs



    Hello,

      We have some new secure IP phones that we are trying to get to go secure with some STE phones that are PSTN connected. We can call the STE phones no problem but when we try to go secure it fails. The manufacturer is telling me they will only work with MGCP gateways and we run h.323. Any experience with this setup?

    Thanks in advance.

    DB:3.52:H.323 Gateway And V.150 zs


    Not to give you a smart answer, but I think the prerequisites are very clear and unequivocal:

    •You must have Cisco IOS Release 15.1(4)M and Cisco UCM 8.6 or later releases installed on your network.

    •You must have the following images and licenses installed and running:

    –The adventerprisek9-mz image is needed for Integrated Services Routers (ISRs)

    –The universalk9-mz image in needed for ISR Generation 2s (ISR G2s)

    –UC and security feature licenses are needed for ISR G2s

  • RELEVANCY SCORE 3.47

    DB:3.47:H.323 Vs Mgcp k8



    Hi ,

    what about Call survivability in h323 and mgcp gateways in both analog as well as IP phones.

    DB:3.47:H.323 Vs Mgcp k8


    Apples and oranges - H323 is a peer to peer protocol where MGCP is controller/controlee. There are ways to have multiple controllers with MGCP and survivable calls - like when ccm fails over or BTS. There is a way to do survivable telephony with H323 using AnnexE, this is supported on the BTS, unsure about CCM.

  • RELEVANCY SCORE 3.42

    DB:3.42:H.323 And Mgcp Gateways - Prioritizing Inbound Calls fj



    I have a T1 PRI terminated in a 2821 router. It has a 100 number DID block. There is a H.323 gateway between the router and a IP Fax server. There is also a H.323 gateway to the call manager and also MGCP gateway to the call manager. I want to use some DID numbers for sending it to fax server and some DID numbers to be sent to the CCM for call processing. In this scenario, is it possible to use H.323 for the fax purpose and H.323,MGCP for calls to the CCM? In other words, can a router have both H.323 and MGCP running together where we can say what protocol needs to do what?

    DB:3.42:H.323 And Mgcp Gateways - Prioritizing Inbound Calls fj


    There is often a debate for which protocol to choose MGCP or H.323? Although it is possible to configure both on the same gateway, this is strictly for MGCP fallback to SRST.

    The key aspects is that with H.323 compared to MGCP, H.323 requires more configuration on the gateway since the gateway must maintain the dial plan and route patterns.

    CallManager only sees the router as one gateway. Calls are sent to the gateway but CallManager cannot specify which port the call is sent to.

    The CallManager does not even know that multiple ports exist on the gateway

    With MGCP however, Cisco CallManager knows and controls the state of each individual port on the gateway.

    MGCP allows complete control of the dial plan from Cisco CallManager, and gives the CallManager per-port control of connections to the PSTN.

    For these reasons call control requires MGCP or H.323.

    Rgds

    Allan.

  • RELEVANCY SCORE 3.36

    DB:3.36:Calls Not Rolling Over To 2nd Member In Route Group cm



    Hi

    I have ccm 3.3.4 on a a non-production server along with a 1760 gateway running 12.3(7)T. The gateway contains a MFT-T1 and a Vic2-2FXO.

    I've created two gateway objects in CCM; an H.323 gateway for the PRI and an MGCP gateway for the two FXO ports.

    I've put the gateways in a route group (PSTN_RG) with the H.323 gateway first and put the route group in a route list (PSTN_RL).

    I've assigned my route list to my route pattern 9.@.

    Normally I would create a separate route group for the analog ports and set PreAt digit discard on that group in the Route List configuration and I would strip the 9 from calls out the PRI in the gateway dial peer. Since I don't have a live PRI in the lab to play with I put everything in the same route group and set the PreAt digit discard setting on the route pattern.

    So what happens is this: I place a call from an IP phone, 9.555-1111 for instance.

    The route pattern strips the 9 and sends the call to the route list.

    The route list sends the call to the first gateway in the route group, which is the H323 PRI gateway.

    The gateway gets the call and looks for an H.323 dial-peer to use. It cannot match the call to an H.323 dial-peer, which has a dest pattern 9.

    I get reorder tone. The call is never sent to the MGCP gateway ports.

    If I change the dest pattern on the H.323 peer to .T so the call will match the H.323 dial peer, the call is sent to the mgcp dial peer and completes successfully.

    It seems that the gateway isn't switching over to the mgcp application if it can't match an h.323 dial-peer. I've tried adding the ccm-manager mgcp-fallback command and the call application alternate MGCPAPP command but they didn't make a difference.

    Is this the expected behavior, am I missing something in the configuration (pasted below), or is this possibly a ios bug?

    gateway configuration

    voice class codec 1

    codec preference 1 g711ulaw

    codec preference 2 g729r8

    !

    ccm-manager mgcp

    ccm-manager music-on-hold

    ccm-manager config server 192.168.2.41

    !

    controller T1 0/0

    framing esf

    linecode b8zs

    pri-group timeslots 1-24

    !

    interface FastEthernet0/0

    ip address 192.168.2.2 255.255.255.0

    speed 100

    full-duplex

    h323-gateway voip bind srcaddr 192.168.2.2

    !

    interface Serial0/0:23

    no ip address

    no logging event link-status

    isdn switch-type primary-ni

    isdn incoming-voice voice

    no cdp enable

    !

    ip classless

    ip route 0.0.0.0 0.0.0.0 192.168.2.1 permanent

    no ip http server

    !

    control-plane

    !

    voice-port 0/0:23

    no vad

    no comfort-noise

    !

    voice-port 1/0

    !

    voice-port 1/1

    !

    mgcp

    mgcp call-agent 192.168.2.41 service-type mgcp version 0.1

    !

    mgcp profile default

    !

    dial-peer voice 3 pots

    destination-pattern 9.

    progress_ind alert enable 8

    progress_ind progress enable 8

    progress_ind connect enable 8

    incoming called-number .T

    direct-inward-dial

    port 0/0:23

    !

    dial-peer voice 1 voip

    destination-pattern 2...

    progress_ind setup enable 3

    voice-class codec 1

    session target ipv4:192.168.2.41

    incoming called-number .T

    dtmf-relay h245-alphanumeric

    no vad

    !

    dial-peer voice 10 pots

    application mgcpapp

    port 1/0

    !

    dial-peer voice 11 pots

    application mgcpapp

    port 1/1

    DB:3.36:Calls Not Rolling Over To 2nd Member In Route Group cm


    Also, check your Service Parameters for "CallManager" on each server. There is a field which says something like "continue routing on busy" which is default to FALSE.... set it to TRUE. I believe this may help.

  • RELEVANCY SCORE 3.35

    DB:3.35:Teho 81



    Hi All,

    Just curious does Tail End Hop off works for H.323 gateways??

    Do i need to setup any Dial Peers on H.323 gateways for TEHO to work??

    Cheers,

    Hunt

    DB:3.35:Teho 81


    this should give you a good idea about TEHO

    https://learningnetwork.cisco.com/docs/DOC-1362

  • RELEVANCY SCORE 3.35

    DB:3.35:Do Anyone Have Solution Or Config Router H.323 To Do Redirect Call p9



    as right now ,we have setup router to do H323 for redirect call back to callmangaer after change called number.

    but after we have tested.it didn't work

    CM -- H.323 (translated called) -- send back to CM -- route call to MGCP gateway.

    do anyone ever config like this scheme?

    the thing that we need to do this callmanager didn't support re-direct call when ipphone A 545-5XXX made call forward to mobile phone , and ipphone B

    323-4XXX call to ipphone A with 5XXX. the call will forward to mobile number that IPPhone A have fixed it. but calling will be 545-4XXX instead of 323-4XXX, after ask cisco for this cause it will use CSS from ipphone A and send out.

    so i try to setup H.323 gateway to fix this issue by send out to H.323 first to translate called ,apply some prefix to route to right gateways,and send back to CM to put call to MGCP and then PSTN.

    regards,

    kanittha

    DB:3.35:Do Anyone Have Solution Or Config Router H.323 To Do Redirect Call p9


    as right now ,we have setup router to do H323 for redirect call back to callmangaer after change called number.

    but after we have tested.it didn't work

    CM -- H.323 (translated called) -- send back to CM -- route call to MGCP gateway.

    do anyone ever config like this scheme?

    the thing that we need to do this callmanager didn't support re-direct call when ipphone A 545-5XXX made call forward to mobile phone , and ipphone B

    323-4XXX call to ipphone A with 5XXX. the call will forward to mobile number that IPPhone A have fixed it. but calling will be 545-4XXX instead of 323-4XXX, after ask cisco for this cause it will use CSS from ipphone A and send out.

    so i try to setup H.323 gateway to fix this issue by send out to H.323 first to translate called ,apply some prefix to route to right gateways,and send back to CM to put call to MGCP and then PSTN.

    regards,

    kanittha

  • RELEVANCY SCORE 3.34

    DB:3.34:Route 911 Calls Out Fxo Port am



    Our remote branch office houses a 1760 router and 12 Cisco 7940's. I am wanting to route all 911 calls out the router's FXO ports, but am unsure how to configure the dial-peer. The gateway is H.323. Does anyone have any examples?

    DB:3.34:Route 911 Calls Out Fxo Port am


    my branch office has a 2811 with an SRST license - i want any 911 or 9911 call to exit the pots line connected to the FXO card.

    thanks.

  • RELEVANCY SCORE 3.34

    DB:3.34:Adding Mgcp Fxs Ports To H323 Gateways p3



    Currently all of our Gateways are H.323 gateways.  Due to a business requirement we are now going to be enforcing our users to use forced authorization codes to place LD calls.  In order to facilitate this on our analog phones it seems the only option is to use MGCP gateways.

    From what I understand we can run multiple signalling protocols on voice gateways.  We have a variety of gateway models but by and large most of these gateways are VG224 models.  I think what I would like to do is keep the current h.323 dial-peer and voice-port settings for the PLAR emergency phones that we have on these gateways and only change the analog phones to MGCP. 

    Most of the route patterns to these h323 gateways look like this... 102[0-5] and then the dial peers on the individual gateways route to the appropriate voice port like this...

    dial-peer voice 1020 pots huntstop destination-pattern 1002 port 2/21

    The Voice port config looks like this...

    voice-port 2/21 timeouts interdigit 7 description tie pr 1520 station-id name PTRM 1020 station-id number 1020 caller-id enable

    My plan is to create the MGCP Gateways in CUCM as wells as the DN's... in this example x1020.  I will then enable MGCP on the gateways.  After that my assumption is that I can individually remove the Voice-port and dial-peer configurations and then add the MGCP dial peers with the port and "service MGCPAPP" commands.

    My other option is to redo the entire gateway at the same time and schedule after-hours down-times to make the change.  I want to avoid this if possible as we have 40+ gateways that need to be changed.

    Basically I just need some guidance or confirmation if my plan will work or if there is a better way to do this?  Are there any caveats or known issues I should look out for when running multiple signalling protocols on the same gateway?

    Thanks,

    Trav Moore

     

     

     

     

     

     

     

     

     

     

  • RELEVANCY SCORE 3.34

    DB:3.34:International Dialing Question fa



    I have an issue where I can dial certain contries, for instance germany at say a number like this, 0114986521111, but not say venezuela, at a number like this, 011584167961111. It has two extra digits. Any ideas? My route patterns are 9.011! and 9.011#. i can't figure out why its not working.CM 6.1 w/h.323 gateways.

    Thanks

    DB:3.34:International Dialing Question fa


    Ok guys thanks for the two posts. that got me digging into it a bit more and I noticed on the isdn debug on the international number that wouldnt go through I was getting "Plan:ISDN, Type:International" and all other calls were "plan:unknown, type:unknown. So I added "isdn map address ^011* plan unknown type unknown" to the serial interface and that fixed the issue. I had noticed during testing that the other PRI's I had at other locations didnt work either but at my locations with Pots lines they had no problems dialing. So I figured something at the telco but this was an interesting fix.

  • RELEVANCY SCORE 3.28

    DB:3.28:Disconnect Delay Occurs After Voip Call Between H.323 Gateways Connected Over Isdn af


    Core Issue
    The network topology is as shown:

    R3640D--(BRI)--(ISDN Simulator)--(BRI)--R3640E
    A VoIP call is established from an H.323 gateway (R3640E) to another H.323 gateway (R3640D). The user expectation is that once the VoIP call is completed successfully, the connection should be terminated and the ISDN link drops by idle-timer (120 seconds) expiration.
    However, the ISDN connection is maintained for 12 minutes. IP packets are seen traversing the ISDN link every minute (like a keepalive).
    Note: This behavior is observed only by 12.3 main. It is not observed in 12.2 main.
    Resolution
    When a call is established across H.323 gateways, where the gateways are the endpoints of the call, the H.225 session remains active for 10 minutes after the call termination.
    This delayed connection termination capability has been added to H.323 gateways. This avoids situations where toggling a single call causes the line to be raised and dropped repeatedly.
    For more information, refer to h225 timeout tcp call-idle (H.323 voice-service).

    DB:3.28:Disconnect Delay Occurs After Voip Call Between H.323 Gateways Connected Over Isdn af

    Core Issue
    The network topology is as shown:

    R3640D--(BRI)--(ISDN Simulator)--(BRI)--R3640E
    A VoIP call is established from an H.323 gateway (R3640E) to another H.323 gateway (R3640D). The user expectation is that once the VoIP call is completed successfully, the connection should be terminated and the ISDN link drops by idle-timer (120 seconds) expiration.
    However, the ISDN connection is maintained for 12 minutes. IP packets are seen traversing the ISDN link every minute (like a keepalive).
    Note: This behavior is observed only by 12.3 main. It is not observed in 12.2 main.
    Resolution
    When a call is established across H.323 gateways, where the gateways are the endpoints of the call, the H.225 session remains active for 10 minutes after the call termination.
    This delayed connection termination capability has been added to H.323 gateways. This avoids situations where toggling a single call causes the line to be raised and dropped repeatedly.
    For more information, refer to h225 timeout tcp call-idle (H.323 voice-service).

  • RELEVANCY SCORE 3.28

    DB:3.28:H323 In Route Lists 93



    I'm just a trying to figure out if I have 2 Route Groups in a Route List and the gateways are H323, how does the Route lists know that the PRI is down? I know that in MGCP it will failover to the other Route Group, but does it do the same for H323 PRI and POTS?

    DB:3.28:H323 In Route Lists 93


    Tdefault behavior in IOS, when there are no valid POTS dial-peers for a call to go out of, is to return "unallocated number" as a cause code to CCM. This can happen even if a t1 goes down. When the T1 goes down, the POTS peer is marked as down, and when CCM sends a call to the gateway, it'll return a UAN

    cause code to the ccm, causing CCM (by default) to stop hunting for other available gateways. There are ways to change the behavior using service parameters in CCM,

    (CCMAdmin|Service|Service Parameter| Stop hunting... flags) but this behavior didn't make sense to me, so we looked for other ways to do it.

    It turns out that you can issue the global command "no dial-peer outbound status-check pots" on the IOS GW, if you're opposed to

    changing the CCM behavior. What this command will do is cause the dial-peer to stay up. IOS will try and route the call, and when the T1 is down, it returns "No circuit available" to CCM. When CCM receives this cause code, it knows there's been a non-user error, and continues hunting, achieving the desired behavior.

  • RELEVANCY SCORE 3.27

    DB:3.27:Ichat 4 And H.323 8d


    Is it possible to make a H.323 conference between iChat 4 and a H.323 hardware device (like a Polycom). if yes, how... ?

    DB:3.27:Ichat 4 And H.323 8d

    Deepest regrets and mea culpa, Ralph!

    Belay that suggestion.I shall renew it at a more opportune time.

    Regards,
    Jim

  • RELEVANCY SCORE 3.24

    DB:3.24:Fax-Relay Fax Pass-Through Integration kj



    Hi,

    I have the task to migrate a network from Fax Relay to Fax Pass-Through. Some gateways are connected via H.323 and some via MGCP to the callmanager 4.2. The dial-peers on the H.323 gateways point to the callmanager. Can I migrate some gateways to Fax Pass-Through without loosing connectivity? How is the call flow in this case? I guess a direct RTP stream between the gateways won't be possible, caused by the different transport types. Will the Callmanager terminate the call flows? If yes, how is the affect to the Callmanager cpu? Sorry for so much questions, but I can't find anything about that.

    Regards

    Andre

    DB:3.24:Fax-Relay Fax Pass-Through Integration kj


    This URl should help you:

    http://www.cisco.com/en/US/tech/tk652/tk777/technologies_tech_note09186a0080159cf3.shtml

  • RELEVANCY SCORE 3.24

    DB:3.24:Dial To Pstn With Out Fac Using Local Route Groups fc



    Hello Guys, I am deploying a national voice network

    using a cluser of call managers (3) an h.323 gateways trough different sites of

    my country.

    The amount of locations I am deploying is about 400. We are using local route groups, as you know it need that all the phones, route patterns, gateways, etc. should be placed in one partition and one css.

    The issue that I am facing right now is that the vip users don´t want to dial fac´s to place a call to the pstn as the mortal ones.  I tried using another partition (VIP) an another css (VIP) but the call request the fac to be established eventhough i have another route pattern in another partition, but still using the h.323 gateway in the common calling search space.

    Is there a work around to solve this issue?

    Regards.

    DB:3.24:Dial To Pstn With Out Fac Using Local Route Groups fc


    Did you remove access to the regular route patterns from VIPs and replaced with new route patterns/partitions?? or just added VIP pt/route pattern to their CSSs?

    FAC and CMC are enabled only in the route pattern.

    If you left access to the regular route patterns and they're a better match, then call will be routed using them.

    If they're equally good matches, then you need to order the CSS to have the VIP partition above the partition from regular route patterns so it is used.

    HTH

    java

    If this helps, please rate

    www.cisco.com/go/pdihelpdesk

  • RELEVANCY SCORE 3.23

    DB:3.23:Ccm And H.323 Gateways 9k



    It appears that a router configured with basic voice ports and dial-peers is termed an "H.323 Gateway", as defined in CallManager. How does this differ to an H.323 Gateway router explicitly configured with 'Gateway' and 'h323-interface voip" commands?

    Does a basic voice port/dial-peer router exchange H.323-style call-setup messages when communicating with CCM (and other VoIP routers which may be contacted to establish call legs)?

    DB:3.23:Ccm And H.323 Gateways 9k


    Hi - in short, yes. H.323 is the signalling used for this setup.

  • RELEVANCY SCORE 3.23

    DB:3.23:Forward Route Pattern When No Aswer/Busy/Down aa



    In CUCM6, is it possible to do send a call towards a route pattern (H.323) and, if the call is busy or the gateway is down, the call should go to a directory number.

    thanks in advance!

    DB:3.23:Forward Route Pattern When No Aswer/Busy/Down aa


    the call should go to a directory number.

    What exactly would be the 'directory number'? Any separate gateway/device? or different DN?

  • RELEVANCY SCORE 3.22

    DB:3.22:H.323 Gateways And Ucm kc



    I inherited an environment that has Unified Communications Manager 6

    .0, but instead of MGCP gateways, the engineers configured it to use h.323 gateways

    is there some advantage to doing this? what are the drawbacks? All the gateways are Cisco

    DB:3.22:H.323 Gateways And Ucm kc


    When configured as H323, the gateway and the server are peers.  You'll have to build and maintain dial peers to pass traffic.  But at the same time, you can share the gateway with other clusters and other H323 services if you wish. 

    With MGCP, the configuration is, in my opinion, easier for most IT staffs to maintain without a lot of special training.  The gateway operates as a slave, and the intelligence is maintained on the servers with this setup.

    H323 may be necessary if you're using one of the more unusual ISDN implementations (ATT has a ton of these like a 4ESS with MEGACOM and they cannot be configured under MGCP), or if you want to use something like NFAS, you have to go H323 because there is no support under MGCP for those options. 

    In some cases, the choice is driven by what you're doing.  Sometimes, it's a personal preference.  But considering what the customer has to do to support the system once built should also be a factor in any optional decisions. 

    Cliff    

  • RELEVANCY SCORE 3.21

    DB:3.21:Capacity Planning - H.323 Gateways c9



    I'm running through some possible changes to my network and there's something I'm not entirely clear about. What would happen if I have a setup like this:

    Route List Generic-RL

    Route Group H.323 gateway

    Route Group Multiple MGCP T1's

    Let's say the H.323 gateway only has 1 PRI. Now I have 23 calls up (the PRI is full). Now I need to send another call out the Route List. Is there anyway to have it go to the second RG instead of sending it to the one for the H.323 and just failing due to lack of capacity?

    Any help would be greatly appreciated.

    Thanks,

    Jim

    DB:3.21:Capacity Planning - H.323 Gateways c9


    Huh, I did not know that. I guess I'd just been assuming it for so long I didn't think it would work. Thanks, that makes things quite a bit easier.

    Thanks,

    Jim

  • RELEVANCY SCORE 3.20

    DB:3.20:Group Pickup Through H.323 d1



    I have been constructing a somewhat strange setup.

    To replace an Aastra MD110, with a large number of analog connections, a CUCM 8.5 has been installed using IAD2430(24fxs) as analog gateways.

    These are not directly supported by CUCM. Therefore the analog extensions has been constructed, using RemoteDestinationProfiles(to hold the local extension (2345), and a remote destination (7652345), pointing to a route pattern (765XXXX), serviced by H.323 gateways.

    Now i'm faced with the task of enabling group pick-up in this scenario.

    Any ideas will be welcome.

    As a side question, I can't find a "midcall feature" guide for phones connected to the IAD2430(24FXS).

    DB:3.20:Group Pickup Through H.323 d1


    Try your luck telling CUCM that the IAD is a VG224.

    I think it supports SCCP mode for the FXS ports.

    With that, all the features can work via dtmf codes.

  • RELEVANCY SCORE 3.20

    DB:3.20:Implementing Voice Gateway On Cucm 10.X 78



    Introduction

    There were few discussion in CSC on implementing voice gateways in CUCM 10.x related to configuration and troubleshooting. To help the users I have decided to write this blog based on my work experience on new CUCM 10.x platform and provide configuration examples for creating MGCP PRI and H.323 PRI on voice gateways and integrating with the PSTN network.
    Configuring MGCP PRI and Gateway on CUCM 10.x
    Here is the link,How to configure MGCP PRI and Gateway on CUCM 10.x ?
    Configuring H.323 PRI and Gateway on CUCM 10.x
    Here is the link, How to configure H.323 PRI and Gateway on CUCM 10.x?

     

    I hope the information in this blog is helpful.

    DB:3.20:Implementing Voice Gateway On Cucm 10.X 78


    Introduction

    There were few discussion in CSC on implementing voice gateways in CUCM 10.x related to configuration and troubleshooting. To help the users I have decided to write this blog based on my work experience on new CUCM 10.x platform and provide configuration examples for creating MGCP PRI and H.323 PRI on voice gateways and integrating with the PSTN network.
    Configuring MGCP PRI and Gateway on CUCM 10.x
    Here is the link,How to configure MGCP PRI and Gateway on CUCM 10.x ?
    Configuring H.323 PRI and Gateway on CUCM 10.x
    Here is the link, How to configure H.323 PRI and Gateway on CUCM 10.x?

     

    I hope the information in this blog is helpful.

  • RELEVANCY SCORE 3.18

    DB:3.18:H.323 Gateway Failover cz



    I have a CUCM 6.0 and 2 ISR2811 acting as H.323 gateways in 2 different branch offices. I have configured a Route Group that with this 2 GWs. The goal is to redirect calls to a second gateway if all the ports in the first one are busy. It's notworking. POTS dial peers are configured in each GW for each dial pattern. I also tried DPs with preference like the folowing ones

    dial-peer voice 7 pots

    !First dial peer, if busy redirects call to dp 71

    preference 1

    destination-pattern 79........

    port 0/1/1

    prefix 9

    !

    dial-peer voice 71 voip

    !Second dial peer, redirects call to second GW.

    preference 2

    redirect ip2ip

    destination-pattern 79........

    session target ipv4:10.0.0.1

    codec g711alaw

    it's not working. Can someone find the reason?

    DB:3.18:H.323 Gateway Failover cz


    Thanks Marwan, Andrew and Christopher for your help. It's working fine right now. The main issue was the flags mentioned by Christopher. As soon as I changed it started working. Thanks again!

  • RELEVANCY SCORE 3.17

    DB:3.17:H.323 And Mgcp Gateway ap



    Dears,

    Greetings , i asked if can deploy H.323 protocol and MGCP protocol in the same gateway using AS5350XM or AS5400XM Voice gateways and 1 TFTP server and 5 call managers 7.1 .

    Regards.

    DB:3.17:H.323 And Mgcp Gateway ap


    With MGCP you are relying on CUCM for call routing and control logic. H323 is all native to the GW.

  • RELEVANCY SCORE 3.17

    DB:3.17:H.323 To Sip Interworking 9f



    I'm looking into interworking a Cisco CallManager solution using H.323 to 36xx, 53xx VoIP Gateways, and Skinny client to IP Phones to a 3rd party solution supporting only SIP.

    Any ideas on a device that could provide this fuctionality cost effectively with minimal delay to Voice traffic... That is available "off the shelf" and prefirably Cisco (for ease of support...)

    DB:3.17:H.323 To Sip Interworking 9f


    my company provides a solution for H323 to SIP interworking. If you are really interested i can have one of our sales engineers contact you. We are still at the teting phase, but i have faith in our product. We inteoperate with major vendors out there, especially with CISCO. This might be what you are looking for. the company website won't be updated until QA gives its blessing to the release, so don't even bother looking there.

    contact me at epatasse@nextone.com

  • RELEVANCY SCORE 3.17

    DB:3.17:Need To Restart Ccm Service To Get Route Patterns Working. sk



    Hi,

    I have problems when adding route patterns pointing to MGCP and H.323 gateways in that I must restart the CCM server. I am running 4.0.2aSR1 with 200-2-6sr5 on 7825s. Is there some know issue here?

    Thx,

    DB:3.17:Need To Restart Ccm Service To Get Route Patterns Working. sk


    I'm having the same problems also running CCM 4.1(2). Also if I modified the name of a partition that is already included within a CSS it will no longer work until I restar CCM Service (problems seems to be happening randomly).

    -Jose

  • RELEVANCY SCORE 3.17

    DB:3.17:Mgcp And H.323 Use At Same Time ? dp



    i have 2611xm voice gateway,

    now i using MGCP.

    do i use mgcp and h.323 with one voice gateway ?

    i think there is not problem if using another route pattern each other.

    DB:3.17:Mgcp And H.323 Use At Same Time ? dp


    thanks about your answer.

    router ios version is 12.3(7)T.

    now, i know that h.323 gateway does not operate.

    strangely , h,323 does not operate.

    below my h.323 configutration,

    plz,check false.

    -------------------------------------------

    p dhcp pool e

    network 192.168.0.0 255.255.255.0

    default-router 192.168.0.1

    option 150 ip 192.168.0.99

    dns-server 210.94.0.73 210.220.163.82

    netbios-name-server 210.96.0.73 210.220.163.82

    voice-card 1

    !

    !

    !

    !

    voice service pots

    !

    voice service voip

    h323

    h245 caps mode restricted

    !

    !

    voice class codec 1

    codec preference 1 g711ulaw

    !

    !

    !

    !

    !

    !

    !

    !

    !

    fax interface-type fax-mail

    !

    !

    class-map match-all class1

    match access-group 101

    !

    !

    policy-map policy1

    class class1

    bandwidth 3000

    random-detect

    random-detect precedence 0 32 256 100

    !

    !

    translation-rule 1

    Rule 0 ^80 0

    Rule 1 ^81 1

    !

    !

    translation-rule 2

    Rule 0 ^90 0

    Rule 1 ^91 1

    !

    !

    translation-rule 3

    Rule 0 ^71 1

    Rule 1 ^70 0

    Rule 3 ^78 8

    !

    !

    !

    !

    interface FastEthernet0/0

    ip address 192.168.1.2 255.255.255.0

    service-policy output policy1

    duplex auto

    speed auto

    h323-gateway voip interface

    h323-gateway voip id Serome Gatekeeper 47 ipaddr xxx.xxx.xxx.xxx 1719

    h323-gateway voip h323-id krisnet1@2004@01:031

    h323-gateway voip tech-prefix 8#

    h323-gateway voip bind srcaddr 192.168.1.2

    ip rtp priority 16384 16383 40

    !

    interface FastEthernet0/1

    no ip address

    duplex auto

    speed auto

    !

    interface FastEthernet0/1.1

    encapsulation dot1Q 1 native

    ip address 192.168.0.1 255.255.255.0

    !

    interface FastEthernet0/1.2

    encapsulation dot1Q 2

    ip address 192.168.2.1 255.255.255.0

    !

    ip classless

    ip route 0.0.0.0 0.0.0.0 192.168.1.1

    ip http server

    ip http path flash:

    prefix-length 29

    ip nat inside source list 1 pool local overload

    ip nat inside source static 192.168.0.50 !

    !

    access-list 1 permit 192.168.0.0 0.0.0.255

    !

    control-plane

    !

    !

    voice-port 1/0/0

    supervisory disconnect dualtone mid-call

    cptone KR

    timeouts interdigit 2

    connection plar opx 122

    caller-id enable type 1

    !

    voice-port 1/0/1

    supervisory disconnect dualtone mid-call

    cptone KR

    timeouts interdigit 2

    connection plar opx 122

    caller-id enable type 1

    !

    voice-port 1/0/2

    !

    voice-port 1/0/3

    !

    voice-port 1/1/0

    !

    voice-port 1/1/1

    !

    !

    !

    !

    !

    dial-peer voice 22 pots

    destination-pattern 155

    fax rate voice

    port 1/1/1

    !!

    dial-peer voice 23 pots

    destination-pattern 117

    port 1/1/0

    !

    dial-peer voice 7777 voip

    destination-pattern 1..

    session target ipv4:192.168.0.99 ----CM IP

    codec g711ulaw

    !

    dial-peer voice 2 pots

    preference 7

    destination-pattern 8T

    translate-outgoing calling 1

    port 1/0/0

    forward-digits all

    !

    dial-peer voice 3 pots

    destination-pattern 8T

    translate-outgoing calling 1

    no digit-strip

    port 1/0/1

    forward-digits all

    !

    gateway

    !

    alias exec c conf t

    alias exec s sh run

    alias exec e exit

    !

    line con 0

    exec-timeout 0 0

    logging synchronous

    line aux 0

    line vty 0 4

    password xxxxx

    login

    !

  • RELEVANCY SCORE 3.16

    DB:3.16:Mgcp And H.323 Gateways Not Working Together ks



    We have a 3640 in Mexico defined to our Call Manager here in Atlanta as a H.323 gateway. Also in the same Call Manager is a T1 PRI defined to our PBX as a MGCP gateway. When people in Mexico call phones connected to our PBX we get one way conversations. We have a couple of other H.323 to MGCP gateways situations that don't work. It seems the two are not compatible. Has anyone else ran into this. If so, what was your solution.

    DB:3.16:Mgcp And H.323 Gateways Not Working Together ks


    Shane thanks for the info. The actual setup is like this

    MexPBX--E1/CAS--3640---h.323--CCM--mgcp---4006/c4gwy--T1/PRI---PBX

    The mex router has 12.1.2T.

    The CallMgr has 3.1.4B.

    The 4006 has 12.2.11T.

    We can here Mex, they can't here us

  • RELEVANCY SCORE 3.16

    DB:3.16:Route Patterns, Css And Partitions With H.323 jx



    Can someone detail from a best practices standpoint how route patterns/CSS/Partitions on the CCM should be configured if using h.323 as gateway protocol?

    In our design, we will have three sites each with PSTN connectivity (HQ houses CCM/VM). Phones need to be seperated by COS (local, local and LD, local LD and International).

    So in the above, do I just create a single route pattern on the CCM for each site (9.@) and then specify the more specific router patterns on the h.323 dial-peers? And then also impliment COS on each h.323 gateway so I can differentiate each COS groups dialing ability?

    Or should we use traditional partitions and CSS's on the CCM, and then for each location setup a local, ld and international route pattern (pointing to the respective h.323 gateway). Then essentially have the same setup on the h.323 from a destination-pattern perspective (that matches route patterns on CCM for each location)? And thus it will be on the CCM that determines which route patterns a phone has access too?

    thanks in advance,

  • RELEVANCY SCORE 3.16

    DB:3.16:Meeetingplace 6.0/Webconferencing Gateways cd



    As far as I know Meetingsplace 6 uses Adobe Flash for Webconferencing.

    So are there gateways for dialing in with SIP and/or H.323 clients into Meetingplace Webconferencing?

    DB:3.16:Meeetingplace 6.0/Webconferencing Gateways cd


    As far as I know Meetingsplace 6 uses Adobe Flash for Webconferencing.

    So are there gateways for dialing in with SIP and/or H.323 clients into Meetingplace Webconferencing?

  • RELEVANCY SCORE 3.14

    DB:3.14:Cucm Failover xf



    I have the following topology.  The Avaya PBX is our primary route to the PSTN.

    My primary route is a SIP trunk between the CUCM and the Avaya PBX.

    I have an H.323 gateway configured between the CUCM and the Avaya PBX.

    I also have an MGCP gateway configuration as a backup to the PSTN.

    I would like to use the SIP trunk as the primary, the H.323 as the secondary, and the MGCP as the tertiary.

    I went into route group configuration, but can't add the gateways as devices.

    Is there a way to add these gateways as devices within the route group?

    Thanks!

    DB:3.14:Cucm Failover xf


    Correct you cannot mix types as my post above points out in our documentation, which likely is why you can no longer add other devices after your MGCP gateway.

  • RELEVANCY SCORE 3.13

    DB:3.13:Can Call Manager Support Network Side Q.931 From H.323 Gateways 3p



    With CCM 3.1, is it possible to support network side E1 Q.931 on a H.323 gateway. This would allow PABXs to connect to CCM, which then allow CCM to act as a tandem exchange.

    DB:3.13:Can Call Manager Support Network Side Q.931 From H.323 Gateways 3p

    The network side support on the H323 gateway

    is transparent to the CCM. They are two seperate call legs.

    1. pots for E1 PRI and

    2. voip call leg to CCM.

    The support for network side PRI is on 2600/3600/3700/7700/AS5300,AS5800

    and its been available since 12.1(3)T and please use

    a more recent image 12.2T train ip plus at the least.

  • RELEVANCY SCORE 3.13

    DB:3.13:Call Blocking Via Cucm dj



    Currently we use translation profiles on our H.323 gateways and apply them to ports/dial-peers to reject specific calls coming inbound from the PSTN. I can see us getting closer to the 15 rule maximum for the translation rules and would like to know what would be the cleanest way to block specific calls from the PSTN by ANI. The only option I could think of was to create a different translation pattern for each with the route option "block this pattern"  We are currently on call manager ver. 8.6.2.23900-10. Thanks!!

    DB:3.13:Call Blocking Via Cucm dj


    James:

    Very interesting, thanks for the reply.  We are running IOS Version 15.2(4)M4 on our 3925Es.  Good to know this version of code supports more rules.  I am still going to look into doing what Jaime and Dennis suggested as I would like to have a central location to manage call blocking.  Thanks!!

  • RELEVANCY SCORE 3.12

    DB:3.12:Creative Way To Route H.323 Calls From/To 2 Routers? mk



    How h323 will be handle 2xPRI with DID attached to different routers in case all 24 channels are taken?

    Basically, if the voice gateways are setup has H.323, and we place PRIs on each router. How can I configure CCM for outbound calls to *both* routers. I know in MGCP its easy with RG and RLs.

    Is it done at the router and some kind of dial peer to push it over to next router if all channels are filled?

    Thanks!

    DB:3.12:Creative Way To Route H.323 Calls From/To 2 Routers? mk


    ahh.. so if they are assigned to a route pattern, then CCM takes it off the RL list as an available gateway. Correct?

  • RELEVANCY SCORE 3.10

    DB:3.10:Consensus On Most Stable Call Control 78



    Which is more stable, an MGCP controlled PRI gateway or an H.323 controlled PRI gateway?

    My experience tells me that H.323 controlled gateways are less problematic than MGCP controlled gateways.

    Has anyone else experienced this or am I off-base?

    Thanks for your feedback.

    DB:3.10:Consensus On Most Stable Call Control 78


    I really dont have lots of issues with either one. They both seem to work fine. I personally prefer H.323 for two reasons. One, it is much easier to troubleshoot call isses. Router debugs are easier for me to read than CCM traces. Two, on most sites, I need some sort of dial peers for the gateway if I fall in to SRST, so I just do H.323.

    I know this does not help much, but it is just my opinion.

  • RELEVANCY SCORE 3.10

    DB:3.10:Voice Gateways, Slt And Pgw 8s



    Hi everyone,

    I dont know if its the right place to start this discussion so forgive me if i am wrong.

    The company in which i am currently wokring is running a call center and IPT as well. i will try to explain the scenario

    voicegateways (AS5400 and AS5350 routers with IOS Version 12.4(15)T7)

    SLT (2651XM with IOS Version 12.2(8)T10)

    PGW (PGW 2200 with SunOS 5.10, MGC)

    I was under the impression that if you want to connect PSTN and a voip network you need a PSTN gateway. So why are we using 3 different types of hardware.

    Follwing are the explainations in a doucment given to me

    the Cisco PGW 2200 provides service providers with the capability to seamlessly route voice and data calls between the PSTN and New  World packet networks.

    Cisco 2611 Signaling Link Terminals (E1 terminates, Take Signaling part and send B n D channels to Voice gateway, Signaling part is resolved by it self and PGW)

    Voice Gateways

    allows terminals of one type, such as H.323, to communicate with terminals of another type, such as a PBX, by converting protocols. Gateways connect an organization’s network to the PSTN

    Any information is much appreciated

    DB:3.10:Voice Gateways, Slt And Pgw 8s


    We indeed have an SS7-ISUP to H.323 conneection, confimred by my team lead. Any other info is appreciated. Also if you could guide me how to check the type of signaling on PGW?

    thanks

  • RELEVANCY SCORE 3.10

    DB:3.10:Pstn Access Code When Centralized Deployment With Ccm 91


    Hi,

    I've got a centralized deployment with CCM6 and two sites. I used two H.323 gateways and set the PSTN access code to 9. I defined my route pattern and put my gateways in a route list. My problem is that I've got most of the calls from site B routed the site A gateway. I've thought of two access codes but what if i've got 100 sites to deal with.

    Please advice,

    DB:3.10:Pstn Access Code When Centralized Deployment With Ccm 91


    this is not a Video over IP question, please use the NetPro categories appropriately

    already answered your other post, please do not duplicate posts

  • RELEVANCY SCORE 3.09

    DB:3.09:Ccm Voice Gw H.323 Failover Issue 3j



    Hi!

    We have a Cisco CallManager cluster with redundant H.323 Voice gateways for

    PSTN connectivity. The CCM version is 4.1(2) and the VG's are Cisco 3745

    routers running 12.3 mainline IP VOICE image. When the PRI links on the

    first VG goes down, the CCM does not forward the calls to the redundant VG.

    The route groups, lists and patterns are configured properly on the CCM.

    Also the dial-peer and POTS config are proper on both the VG's. The problem

    is only with outbound calls. We tried upgrading the IOS on the VG, still no

    luck.

    Is this problem related with the IOS feature set, like the IP Plus? Pls

    suggest...

    Rgds.

    Sri,

    DB:3.09:Ccm Voice Gw H.323 Failover Issue 3j


    go to CCM Admin page Service Service Parameters choose CallManager Service

    look for Clusterwide Parameters (Route Plan) Stop Routing On User Busy

    Flag* Set it to FALSE at the same place : Stop Routing on Unallocated Number

    Flag* set it to FALSE update

  • RELEVANCY SCORE 3.08

    DB:3.08:Mgcp And H323 Call Preserve zc



    Hi everyone.. Iam new to this community. I have a doubt which Iam not able to clear about call preserve in MGCP and H.323.

    In Gatekeepers and Gateways text book I read that "MGCP supports call survivability for analog and T1 CAS calls, and MGCP does not support

    call survivability for PRI and BRI calls " and "H.323 supports PRI call preservation, and H.323 does not support call survivability by default". Can anyone plz explain this in detail as Iam confused whether they preserve call or not... Thanks

    DB:3.08:Mgcp And H323 Call Preserve zc


    PRI calls will only be preserved with H323 by enabling call preserve service parameter and adding call preserve command on the GW. MGCP is backhauled to CUCM so calls are not preserved with MGCP. T1/E1 cas are not preserved with either protocol.

    HTH,

    Chris

  • RELEVANCY SCORE 3.07

    DB:3.07:Dual Default Gateways On 2611 xj



    I have a 2611 with two ethernets to two different gateways. It currently has 10.191.1.253 for default gateway, and the other gateway is 10.192.1.253. I want it to automatically switch to 10.192.1.253 when 10.191.1.253 is unavailable.

    I have the following:

    ip route 0.0.0.0 0.0.0.0 10.191.1.253 1

    ip route 0.0.0.0 0.0.0.0 10.192.1.253 5

    But that doesn't seem to be working.

    Any help is appreciated.

    Thanks,

    --H

    DB:3.07:Dual Default Gateways On 2611 xj


    It will not work like this. Both routes are the same but I suppose you mistyped an IP adress. You need to run a routing protocol to make the router adjust to changes in the network. Try to install the primary route throug EIGRP or so and define a floating static for the second one, i.e. with a distance that is higher than the EIGRP distance to the primary gateway:

    ip route 0.0.0.0 0.0.0.0 10.192.1.253 130

  • RELEVANCY SCORE 3.07

    DB:3.07:Displaying Caller Id Name On Ip Phones 9z



    Is there a way to display caller ID name incoming in a facility IE on a PRI from carrier on an IP phone? The gateways are 3845 and 28xx running 12.4.9T, configured for h.323.

  • RELEVANCY SCORE 3.07

    DB:3.07:Teho, Toll Bypass With Cucm 7.1 And H.323 Voice Gateways az



    I'm currently trying to setup teho (toll bypass) with a Call Manager 7.1 and 4 h.323 gateways, I need to do a lot of digit manipulation so I decided to make my voice gateways H.323 (instead of MGCP), but now it's turned out to be a real problem to get teho to work; because , how will my cucm or gateways know that another gateway is not available, or has no free pstn lines?

    With MGCP you can create a route list, but with H.323 it just sends the call to another gateway without doing a follow up. Does anyone have any suggestions? (I do not have a gatekeeper) can gw be h.323 and mgcp?

  • RELEVANCY SCORE 3.05

    DB:3.05:Configuring Fsx(For Fax) In Cucm For H.323 Gateway fj



    Hi All,

    I got excellent information in our Cisco support community for configuring FSX in CUCM for H.323 gateway but having some doubts in that. Below are the in and out bound call flow.

    Out bound call for FXS port

    FXS---Dial peer VOIP---CUCM---Route pattern---BRI-PSTN.

    In bound call for FXS port

    BRI---Dial peer VOIP---CUCM---Route pattern---FXS.

    Both FXS and BRI are in same H.323 gateway. I guess, I need to create a  dedicated route pattern(FAX ext 1000) for the FXS port and targeted to H.323 gateway. Correct me if i am wrong.

    thanks,

    Murali.

    DB:3.05:Configuring Fsx(For Fax) In Cucm For H.323 Gateway fj


    Hi All,

    I got excellent information in our Cisco support community for configuring FSX in CUCM for H.323 gateway but having some doubts in that. Below are the in and out bound call flow.

    Out bound call for FXS port

    FXS---Dial peer VOIP---CUCM---Route pattern---BRI-PSTN.

    In bound call for FXS port

    BRI---Dial peer VOIP---CUCM---Route pattern---FXS.

    Both FXS and BRI are in same H.323 gateway. I guess, I need to create a  dedicated route pattern(FAX ext 1000) for the FXS port and targeted to H.323 gateway. Correct me if i am wrong.

    thanks,

    Murali.

  • RELEVANCY SCORE 3.05

    DB:3.05:Cucm Configuring Calling Party Selection For Two Different Destinations 9c



    Hi,

    I have several voice gateways (2821, 2851, 3825) running IOS with SPSERVICESK9 feature set, version  12.4(24)T1 (or higher), connected to CUCM version 7.1.5.31900-3.

    The gateways are configured as H.323 gateways on the CallManager,

    and are configured as PSTN gateways with PRI interface and SIP gateways to MS MOC.

    In forwarded calls to PSTN I need to send "Last Redirect Number (External)" calling party number,

    in forwarded calls to MS MOC I would need to send "Originator" calling party number (as it is an internal call).

    Is there any way how to configure CUCM and the gateways to to use different "Calling Party Selection" for calls going to different destinations?

    Regads     Frantisek Opravil

    DB:3.05:Cucm Configuring Calling Party Selection For Two Different Destinations 9c


    Hi,

    thank you for your reply.

    I have red references from your reply and I conclude, that

    If there are forwarded external calls in the CUCM cluster, and

    some are forwarded to PSTN network (and PSTN provider does not provide "no screening CLI") and

    some are forwarded to MOC (Microsoft Office Communicator, withing our company, i.e. they are internal calls)

    then I need 2 different "calling party selection" in the CUCM Gateway configuration     "last redirect number (external)" for forwarded calls outgoing to PSTN, and

         "originator" for forwarded calls outgoing to MOC

    and it cannot be configured concurrently for a single gateway;

    I need 2 gateways (to PSTN and to MOC) or a gateway to PSTN and SIP trunk to MOC.

    Regards          Frantisek Opravil

  • RELEVANCY SCORE 3.05

    DB:3.05:Compatibilidad H.323 xc


    Hola! quería preguntarles si algunos sabe si los gateways que ahora son small business de cisco (antes linksys), son compatibles con H.323. Estuve buscando por todos lados y no puedo encontrar la data. Gracias!

    DB:3.05:Compatibilidad H.323 xc

    Hola! quería preguntarles si algunos sabe si los gateways que ahora son small business de cisco (antes linksys), son compatibles con H.323. Estuve buscando por todos lados y no puedo encontrar la data. Gracias!

  • RELEVANCY SCORE 3.04

    DB:3.04:Ccm With H323 Gw fp



    hi

    in case when using CCM with an H323 gateway, as H.323 gateways have the intelligence to place and receive calls, then how should we configure the whole dial plan (call routing, privilages,....) on this H323 or on CCM

    if on H323 then do we have to configure on CCM any route patterns, lists, partitions and CSS's,...?

    DB:3.04:Ccm With H323 Gw fp


    Technically 9T should work, but it may cause interdigit timeout issues with certain numbers such as 911, 10 digit local, 11 digit long distance etc. So its recommended that you separate these patterns out as follows.

    dial-peer voice 1 pots

    destination-pattern 911

    port 0/0/0:23

    forward-digits all

    num-exp 9911 911

    dial-peer voice 2 pots

    destination-pattern 9[2-9]..[2-9]......

    port 0/0/0:23

    forward-digits 10

    dial-peer voice 3 pots

    destination-pattern 91[2-9]..[2-9]......

    port 0/0/0:23

    forward-digits 11

    dial-peer voice 4 pots

    destination-pattern 9011T

    port 0/0/0:23

    prefix 011

    You are right you will need to create route patterns in CCM and also on the router. You dont typically need to create COR in routers, if you have CSS restrictions on CCM.

    HTH

    Sankar

    PS: please remember to rate posts!

  • RELEVANCY SCORE 3.04

    DB:3.04:Sip H.323 98



    Hi,

    are the h.323 and SIP protocols supported by the 7975G phone?.    Which Cisco IP phones support both SIP and H.323.

    Thanks

    DB:3.04:Sip H.323 98


    7975G only support Skinny Client Control Protocol (SCCP) and Session Initiation Protocol (SIP) with Cisco call control

    http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/ps8538/product_data_sheet0900aecd8069bdb7.html

    As far as i know, 7905G used to support H323(End of Support though. http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/phones/ps379/prod_end-of-life_notice0900aecd80339700.html)

    HTH

    Divin

    PS:Rate useful posts!.

  • RELEVANCY SCORE 3.04

    DB:3.04:Cube Situation ss



    Hello!

    I am about to make an IP telephony installation (CUCM 7.0) with 3 telecom providers as gateways to PSTN.

    Two of them will be connected to one Cisco router (2911) where I will install CUBE (IP to IP gateway). My side will have H.323 signaling, while connection to the providers will  be SIP. Third provider will be connected via PRI interface on 1760 router, signaling to CUCM also H.323.

    First 2 telecom providers will provide me with 10 voice channels each on their connections.

    Third telecom provider (PRI) will provide me with 30 voice channels.

    3 different group of users will be created, each of them with one of providers as default gateway for outside calls.

    Here is my problem, if for example first group of users that has all of theirs 10 channels reserved (taken) tries to make a call, will this call be transfered to the second provider, and if this is taken as well, will it be forwarded to PRI link? Of course I will configure route lists and route groups that all users have primary, secondary and tertiary PSTN gateway. two providers on CUBE will be distinguished via prefixes and with dial peers calls will be forwarded to specific provider.

    Do you have some suggestions how to configure this, will this work, is my construction viable and what do I have to pay special attention to?

    Thank you!

    Best regards,

    Darijo

    DB:3.04:Cube Situation ss


    First, from a design perspective, I'd recommend doing SIP between CUBE and CM, so that you can do early offer forced and don't have to require an MTP for calls out the trunk.  With H323 you need to require an MTP for outbound fast start to get early offer to your provider.

    You can limit the amount of calls to each provider in several ways, in my decending order of preference:

    -Use Call Admission Control (locations) on CM to only allow certain amounts of concurrent calls/bandwidth to each provider.

    -Use the 'max connections' command on the CUBE dial-peers to limit the amount of active calls out each peer.

    -Have the provider reject calls over a certain value with a certain response, and get the gateway or CM to hunt off that and to the next applicable match.

    In all of those scenarios, you'll want to make sure you have each trunk listed in a separate route group in CM.  If you're using the same CUBE for both SIP trunks for hunting the calls to each provider individually, it will get a little trickier, but still can be done.

  • RELEVANCY SCORE 3.04

    DB:3.04:From The Beginner 9c



    Hi,

    I'm trying to learn about VoIP and read some about VoIP signalling protocols (H.323, SIP, MGCP etc.). Everything is OK when reading about a protocol but when it comes to integrate things in my mind I'm really confused. I read about SIP protocol and configs and it seems OK, and same for the H.323. But when I read about Cisco AVVID I see nothing about H.323 gateways or SIP proxies, it's all about CallManager. I've problems with integrating concepts about these protocols and CallManager. Could anyone explain the relation between these concepts or advice some documents?

    Regards.

    DB:3.04:From The Beginner 9c


    These are true statements. The designed solution is determined by the customer requirements. Each of these solutions has it's own features, limitations and dependencies. Basically, what you want to do determines your options on how to do it.

  • RELEVANCY SCORE 3.03

    DB:3.03:H.323 Cucm Support fm



    Hi,

    I have a working call manager version 4.1 and i will upgrade soon to version 6.1, I just need to know if I have to upgrade cisco h.323 router gateways and VG200 which are contacted through H.323 only.

    DB:3.03:H.323 Cucm Support fm


    what about third party H323 gateways?

  • RELEVANCY SCORE 3.03

    DB:3.03:Ccm 3.2 And H.323 Gatekeeper Issues xc



    Hi,

    I'm configuring voice calls between FXS ports on a 2621-XM at 8 sites and IP Phones at the head office.I've configured POTS and VoIP dial peers and H.323 Gateways(GW) on the routers. Also configured a H.323 Gatekeeper(GK) on a 3640. The CCM and the 2621 GW register with the GK fine. I've created a H.323 GK on the CCM pointing to the IP Address of the 3640 GK using Anonymous Device. I've also created a route pattern pointing to remote sites.

    I can place calls from the FXS phones to the IP Phones in the head office but get a fast busy tone when I phone from the IP Phone to the FXS phone in any of the offices.

    I could also create 8 GW on the CCM but I rather create only one GK on the CCM so that the CCM uses the GK to call the remote sites. Any ideas on what might be wrong? I notice the CCM 3.3 has the option of trunk but 3.2 does not have that option. I think by creating the anonynous device GK on 3.2 CCM, I'm actually creating a trunk. Any thoughts? Replies will be highly appreciated. Thanks in advance.

    DB:3.03:Ccm 3.2 And H.323 Gatekeeper Issues xc


    Hi!

    Thanks for the reply. This document is helpful to give a better understanding. However, it talks about 3.3 and 4.1. In 3.2, there's no option to create a trunk device and my understanding is that the trunk functionality is replaced by anonymous device and then creating a route pattern to route to an anonymous device. The IP address of the anonymous device changes to reflect the destination gateway and the gatekeeper provides that address. Is that correct? I have set a similar scenario but the I can't places calls from IP phones to analog phone using the GK. Thanks for your help.

  • RELEVANCY SCORE 3.02

    DB:3.02:Changing Disconnect Cause On H.323 Gateway In Cas Of Isdn Failure d3



    Hi

    We're running CUCM 7.1.2 and H.323 Gateways in each office. We are already using the HQ Gateway as backup in case of an ISDN failure in a remote office. By default the H.323 Gateways are sending the disconnect cause of "1" (Hex 81, unallocated or unassigned number, see debug below) in case of an ISDN Port failure.

    Is there a command to change the disconnect cause (the H.323 one, not the ISDN disconnect cause) to something more accurate like Hex A1 "circuit out of order" or Hex A6 "network out of order"?

    Any suggestions are welcome.

    Many thanks in advance

    Regards

    Stefan

    DB:3.02:Changing Disconnect Cause On H.323 Gateway In Cas Of Isdn Failure d3


    Hi Nick

    You're my H.323 gateway hero ;-)

    With the "no dial-peer outbound status-check pots" the gateway sends a "Requested channel not available" (44, Hex AC) back to the CUCM when the ISDN interface is down. Which seems to be more appropriate to me.

    189361: Jul 31 08:26:13 CEST: //1423/00760A0A2700/H323/cch323_h225_send_release: Cause = 44; Location = 0

    Many thanks for your support and have a nice weekend

    Regards

    Stefan

  • RELEVANCY SCORE 3.01

    DB:3.01:Single Did For Voice And Fax 33



    I am looking for a solution where I am able to have a single DID number that can be used for both voice and fax calls. Essentially, the system needs to detect the signaling from the fax call and route it to e-mail rather than ringing the phone. The same DID needs to route voice calls to the phone, however. I am currently using Comm Manager 7.x and H.323 gateways. I've been told that there is a solution from I3 that can do this. Can we do it with Cisco?

    DB:3.01:Single Did For Voice And Fax 33


    Hello,

    The new TCL fax detect script that has T.38 support is version 2.1.2.3 and it is downloadable from the CCO software center for valid cisco.com users. Updated documentation for the fax detect script and its T.38 support is available for IOS releases 12.4 and 12.4T.

    Prior to TCL 2.1.2.3

    Prior to April 2009 and the release of TCL version 2.1.2.3, Cisco did not provide a T.38 fax detect script similar to the T.37 fax detect script that is available for download on CCO. Therefore, the alternatives for customers looking for a T.38 fax detect script prioir to version 2.1.2.3 are as follows.
    1) If the customer is proficient at TCL then the existing T.37 fax detect script can be altered for T.38 fax relay. The one drawback to this approach is that now the customer has to support the script as it has been altered. 2) Customer can contact Cisco Developer Services to alter/create the T.38 fax detect script. Cisco Developer Services will now support the script but there is a charge for this service.
    Be aware that fax detect is also referred to as Single Number Reach (SNR) because it provides the ability for fax and voice calls to share a single inbound DID.You should also be aware that TCL scripts such as fax detect are not supported on MGCP voice gateways.

    Hope this helps.

    Cheers,

    Jose

  • RELEVANCY SCORE 3.01

    DB:3.01:How To Create Dial-Peer That Is Similar To Route Pattern Like This? zk



    I have route pattern font color=red0[1234]/font for urgent calls with MGCP gateways.

    Now I have changed this numbering plan

    from font color=red0[1234]/font to font color=red#.10[1234]/font with Predot Discard Digits for MGCP gateway.

    But I also have H.323 gateways.

    What right dial-peer should I use in this position?

    The old dial-peer is below.

    b

    dial-peer voice 5 pots

    tone ringback alert-no-PI

    destination-pattern 0[1234]

    no digit-strip

    direct-inward-dial

    port 0/0/0:15

    /b

    How I should change it to receive same result that I described for Route Patter?

    DB:3.01:How To Create Dial-Peer That Is Similar To Route Pattern Like This? zk


    You can try it - but I believe if you try using the terminator in your dial string, it may not act accordingly.

    In this case we're changing the default terminator so that you can use it in your dial strings. I haven't tried it, so you may want to try it first without the change. You wouldn't need your command, because # is the default terminator.

    hth,

    nick

  • RELEVANCY SCORE 3.00

    DB:3.00:Cucm 8.X Import/Export Cannot Import Route Groups (Invalid Qsig Configuration) s8



    Hi,

    When trying to import route groups via the BAT import/export utility I get  the follwoing message in th error log "Invalid QSIG configuration: at least one QSIG variety must be declared". The route groups have been exported from another cluster on the exact same version on CUCM. At the moment the route groups are set to a mixture of 'Mixed non-QSIG' and 'H.323 gateways' I have tried changing them around with both of these options but I still get the same failure message.

    Thanks in Advance

    Andy

    DB:3.00:Cucm 8.X Import/Export Cannot Import Route Groups (Invalid Qsig Configuration) s8


    Hi,

    I figured this out in the end, if anyone wants to know. I found that the gateway.csv although saying it had completed successfully had not imported the H323 gateways so it was causing the route groups to throw this error as there were no gateways to add to the route groups.

    Thanks

    Andy

  • RELEVANCY SCORE 3.00

    DB:3.00:Ccm Fail Over Call Persistance Through Gateways... jf



    I am testing various Cisco PRI gateways in a CCM cluster (2x MCS7835 running v3.09) to understand call persistance during CallManager fail over - i.e. for a 24*7 voice environment. I have the following results when failing the "active" CCM subscriber (so that IP Phones gateways are supported by the publisher):

    6624 - IP Phone to/from analogue Calls remain up with speech path

    VG200 (MGCP) - IP Phone to/from analogue calls remain up with speech path

    IOS H.323 - calls are dropped (as expected as the gateway re-homes onto the other CallManager!)

    DE-30+ - IP Phone ingress/egress calls lose speech path

    6608-E1- IP Phone ingress/egress calls lose speech path

    Now drop and re-establish all the calls, when the failed CCM (subscriber) is restored and the IP Phones and gateways re-register there is no interruption in the speech paths!!!! (Execpt for the IOS H.323 mode gateways which will drop the calls, again)

    My real question is, have I missed some fundamental config or are there resticted gateways that support call persistance during CCM failovers? I had understood that Skinny based gateways maintained calls during CCM failovers - my testing shows this not to be the case!

    Lastly, I had heard that there was a coming CCM v3.1 or IOS feature for H.323 gateways that will preservce calls during failovers - does anyone know if this is the case?

    Regards... Phil.C

    DB:3.00:Ccm Fail Over Call Persistance Through Gateways... jf


    Phil, your findings seem correct to me for CallManager 3.0(x) releases of software.

    In CallManager 3.1, all the Skinny gateways such as the 6608-E1, 6624, and DE-30+ will actually use the MGCP protocol to signal to CallManager. The speech path will survive if the CallManager they are registered to fails.

    Also note that certain IOS platforms, like VG200, 2600, 3600, as well as a couple others, will have newer IOS MGCP software that will also be able to speak MGCP with CallManager 3.1, even with other interface types in addition to FXO/FXS, such as PRI and CAS. So they will be able to have the speech preserved on a CallManager failure as well.

  • RELEVANCY SCORE 3.00

    DB:3.00:Skype As Sip Ua 3x



    Hi,

    We are planning to route our international calling traffic via Skype. Our existing setup is as below

    Cisco CallManager 7.1.5 with H.323 gateways

    Gateways 2951, 2921, 2911, 2901

    My question is whats is CUBE? Do i need CUBE for it, Our existing hardware support CUBE.

    DB:3.00:Skype As Sip Ua 3x


    If this is the only punishment then i am going to use CUBE. Will purchase later .

    Thanks aokanlawon

  • RELEVANCY SCORE 3.00

    DB:3.00:Which Isdn Services Can I Route Via H.323 xd



    Hi,

    I would like to know, if I can route all ISDN services (voice, fax, data) via a H.323 trunk and if the trunk is transparent to the Service Indicator (SI). I plan to use a 3845 as H.323/PRI Gateway.

    regards

    Carsten

    DB:3.00:Which Isdn Services Can I Route Via H.323 xd

    Hi,

    yes - you will be able to route all the specific services (voice, fax, data) via H.323. For modem calls via H323 you can configure MoIP (Modem over IP) with "clear-channel" codec (pure 64Kbps).

  • RELEVANCY SCORE 3.00

    DB:3.00:Call Transfer From H.323 Device px



    I am sending a call to an IP-IVR (non-Cisco) which requires transfer to a CallManager SCCP phone after treatment.

    Here is our current configuration:

    MGCP-controlled PRI on Cisco 3845 ISR, with CallManager 4.1(3). IVR is configured as an H.323 Gateway with route patterns/route groups/route lists appropriately configured to get calls to the IVR. We are not using gatekeepers.

    I can set the H.323 transfer method in the IVR to either hookflash, H.450.2, H.450 Join, or MediaRedirect.

    It is my understanding that CallManager doesn't support the H.450.2 method.

    Any suggestion for getting this done?

    Thanks!

    DB:3.00:Call Transfer From H.323 Device px


    Correct CallManager supports Empty Capabilities Set (ECS) not H.450.

    Media Termination Point Required option in H323 GW config in CCMAdmin is needed if the H.323 clients and H323 devices do not support the H.245 Empty Capabilities Set.

    HTH

  • RELEVANCY SCORE 2.99

    DB:2.99:H.323 Trunk-To-Trunk Routing With Cucm 9-10 - Best Way To Program? jp



    Looking for some expert guidance on the best way to implement trunk-to-trunk call routing in CUCM.  By "best" I mean user friendly/manageable/efficient etc...

    Here's the application:  Inbound TDM calls will be hitting a large, H.323 gateway with 10-digit DNIS.  I need CUCM to recognize the dialed number, then immediately route the call back out one of several smaller, H.323 gateways (depending on the dialed number) while passing the original dialed number to the system at the receiving end of the call.

    The inventory of dialed numbers is large, and routing changes to individual numbers will be made by non-technical staff; so I would prefer to maintain the routing rules in CUCM rather than in the gateways.

    One other layer of complexity:  I need the ability to route the calls to destinations that may include only a single gateway, or span multiple gateways.

    Any suggestions for a best way to implement this in CUCM would be greatly appreciated.

    DB:2.99:H.323 Trunk-To-Trunk Routing With Cucm 9-10 - Best Way To Program? jp


    Simply make sure whatever dialed digits you get from GW A, via the inbound CSS, match a RP in CUCM which might point to an individual GW or a RL.

    If necessary, adjust the dialed digits by adding/removing digits.

    HTH java if this helps, please rate www.cisco.com/go/pdihelpdesk

  • RELEVANCY SCORE 2.99

    DB:2.99:Getting Gateways,Route Partitions, Etc From Axl fa



    I am looking for a way to extract all of the h323/mgcp gateways, route groups, route patterns, and route lists from my 4.1 callmanager. Yet I don't see any listXXX command relevant to these items. Is there another way?

    DB:2.99:Getting Gateways,Route Partitions, Etc From Axl fa


    We just had a bit of a discussion about using AXL vs direct database access here:

    http://forum.cisco.com/eforum/servlet/NetProf?page=netprofforum=IP%20Communications%20and%20Videotopic=IP%20Phone%20Services%20for%20DevelopersCommCmd=MB%3Fcmd%3Ddisplay_location%26location%3D.1dd89d0f

    Not sure if that's of any interest, but some light reading on some of the pros and cons!

    There are some AXL queries that might be of use to you, such as:

    listDeviceByNameAndClass

    %

    Gateway

    where class = Gateway are gateways, though they may be FXS ports as well

    Other queries you might want to check out are:

    listRoutePartitionByName

    or

    listRoutePlanByType

    Route

    or

    executeSQLQuery

    (free-form sql query to get customized result based on the sql query you pass in...)

    The devices (gateways) are the easiest to get. Take a look at the device table in CallManager's CCM03xx database (where xx is the highest-numbered database currently in the CallManager). Here's a sample SQL query:

    select * from device, typeclass, typeproduct, typemodel

    where device.tkclasss = typeclass.enum

    and device.tkproduct = typeproduct.enum

    and device.tkmodel = typemodel.enum

    Devices with a tkClass of 2 are gateways, though as mentioned they may be FXS ports as well, so you may want to filter for that. See the values in the product and model fields for additional device details.

    Routelist, RouteGroup, and Numplan will also be useful tables...

    Just remember that when forming your own queries, if a column begins with "fk", then it links to a pkid in the table, so fkdevice links to the pkid column in the device table; and if a column begins with tk, then it links to an enum in the table (with a type prefix), so tkproduct links to the enum field in the typeproduct table.

  • RELEVANCY SCORE 2.98

    DB:2.98:Cvp W/ Cups k8



    All of my CVP installs have been either H.323 or SIP with without CUPS. So, I have a question.

    On the initial setup, when configuring CUPS with the static routes, I am confused on what I do for the static route for the following components:

    - Labels - If my cvp label is 1111111111, I'll create a static route of 111* but where do I point that to? It says to point it to the VXML Gateway, but in a large branch environment I may have 100 voice gateways. Do I pick a few gateways to use for IP originated calls and alow SIP to always use the originating gateway?

    - Same thing goes for the error and ringback. lets say they are the default of 91919191 and 92929292. Where do I point these to as well in a large branch deployment?

    - tx

    DB:2.98:Cvp W/ Cups k8


    No worries this design was about a week of battle between our Solutions Architect, Myself and one of Our IOS/CCM Gurus. Brains combined we came up with this and it works great!

    Cheers,

    Chad

  • RELEVANCY SCORE 2.98

    DB:2.98:Clarification On Outbound Callerid Configuration 1s



    I need some guidelines on how to set outbound callerID in this environment:

    1) Call Mgr 7.1.5

    2) H.323 gateways

    3) A mix of phones with DIDs who need to mask the DID and other phones who need to mask the main number.

    I'm trying to minimize the number of RG/RL/Route Patterns. Can I do this with a single RL/RG?

    From the route-list detail:

    Use Calling Party's External Phone Number Mask:Default

    Calling Party Transform Mask: 222-555-1234

    Do I hard set the Use Calling Party's External Phone Number Mask to "on" and if there is no Mask on the phone line, it will use the main number mask of 222-555-1234?

    DB:2.98:Clarification On Outbound Callerid Configuration 1s


    I need some guidelines on how to set outbound callerID in this environment:

    1) Call Mgr 7.1.5

    2) H.323 gateways

    3) A mix of phones with DIDs who need to mask the DID and other phones who need to mask the main number.

    I'm trying to minimize the number of RG/RL/Route Patterns. Can I do this with a single RL/RG?

    From the route-list detail:

    Use Calling Party's External Phone Number Mask:Default

    Calling Party Transform Mask: 222-555-1234

    Do I hard set the Use Calling Party's External Phone Number Mask to "on" and if there is no Mask on the phone line, it will use the main number mask of 222-555-1234?

  • RELEVANCY SCORE 2.97

    DB:2.97:H.323 Aaa. ad



    Hi,

    We are using both Cisco Gatekeeper Gateways in our network and just wondering how will I be able to control the authentication of gateways that are registered onto my gatekeeper. Do I need additional software just to control gateway access or am able to do it with the gatekeeper ?

    Thanks,

    Ron Tan

  • RELEVANCY SCORE 2.97

    DB:2.97:Call Preservation 89



    What is the difference w.r.t to Call Preservation for H.323 MGCP Gateways ? Which Gateway will preserve an on-going call, MGCP or H.3232 if the WAN link fails ?

    DB:2.97:Call Preservation 89


    Rob,

    Thanks for your help. Happy weekend !

    Regards,

    Ashok.

  • RELEVANCY SCORE 2.96

    DB:2.96:Adding H.323 Gateway Into Route Grout fc



    At CCM 4.2 - is it possible to adding h.323 gateway into Route group/list? When I try it, at "Insert Route Group" page I dont see any of my installed (and working) gateways. Why?

    Regards,

    DB:2.96:Adding H.323 Gateway Into Route Grout fc


    This is a great document:

    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/3_1_1/ccmsys/a03rp.html

    And this one too:

    http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803ed699.html#wp1311511

    Take a quick read and if you still need any help please let us know. We will be more than happy to assist.

    Ajaz

  • RELEVANCY SCORE 2.96

    DB:2.96:Unity 4.0(4) Fax-To-Mail Application m9


    Hi.

    Does FAX-ON-RAMP works with MGCP gateways or with H.323 only? I have all my gateways in MGCP mode and I would like to incorporate the Fax-To-Mail application using the Gateways and the IP Fax Service on Unity 4.0

    TIA,

    -Jose

    DB:2.96:Unity 4.0(4) Fax-To-Mail Application m9


    Hi Jose,

    unfortunately T.37 (onramp or offramp) only works with H.323. It may be possible to run H323 on just one or two POTS circuits (however many circuits) you want and MGCP for the rest, although I haven't tested that extensively.

  • RELEVANCY SCORE 2.96

    DB:2.96:Asa And H.323 cp



    Hello, colleagues,

    I have some problems with configuring H.323 on ASA 5505 Version 7.2(1)

    At the inside network there is a H.323 gateway with address 192.168.1.8

    Static NAT is made that outside address X.X.X.X is forwarded to the inside address:

    static (inside,outside) X.X.X.X 192.168.1.8 netmask 255.255.255.255

    ACL allow to pass traffic from particular H.323 gateways at the outside network.

    There are inspect commands:

    policy-map global_policy class inspection_default  inspect h323 h225   inspect h323 ras

    I run :

    debug h323 h225 asndebug h323 h225 eventdebug h323 h245 asndebug h323 h245 event

    I see that ASA accept H.225 SETUP message, forward it to the inside address, received CALL PROCESSING from inside gateway, but don't forward it to the outside originating gateway.

    After a period of time the outside originating gateway sends releaseComplete.

    How to fix this issue? Is it possible that H.323 gateway at an inside network behind ASA works correctly with outside gateways?

    I attached the debug output from the ASA.

    Thanks

    DB:2.96:Asa And H.323 cp


    Thanks for letting us know. If it broke even after the upgrade - we were at a loss.

    -KS

  • RELEVANCY SCORE 2.95

    DB:2.95:Can Call Manager Support Network Side Q.931 From H.323 Gateways 8p



    With CCM 3.1, is it possible to support network side E1 Q.931 on a H.323 gateway. This would allow PABXs to connect to CCM, which then allow CCM to act as a tandem exchange.

    DB:2.95:Can Call Manager Support Network Side Q.931 From H.323 Gateways 8p

    The network side support on the H323 gateway

    is transparent to the CCM. They are two seperate call legs.

    1. pots for E1 PRI and

    2. voip call leg to CCM.

    The support for network side PRI is on 2600/3600/3700/7700/AS5300,AS5800

    and its been available since 12.1(3)T and please use

    a more recent image 12.2T train ip plus at the least.

  • RELEVANCY SCORE 2.95

    DB:2.95:Pstn Access Code When Centralized Deployment With Ccm 9k


    Hi,

    I've got a centralized deployment with CCM6 and two sites. I used two H.323 gateways and set the PSTN access code to 9. I defined my route pattern and put my gateways in a route list. My problem is that I've got most of the calls from site B routed the site A gateway. I've thought of two access codes but what if i've got 100 sites to deal with.

    Please advice,

    DB:2.95:Pstn Access Code When Centralized Deployment With Ccm 9k


    Define your route groups

    SiteA_RG = RTRA

    SiteB_RG = RTRB

    Define your route list

    SiteA_RL = SiteA_RG

    SiteB_RL = SiteB_RG

    SiteA_RL = SiteA_RG, SiteB_RG

    SiteB_RL = SiteB_RG, SiteA_RG

    Create your call route partitions

    SiteACalling_PT

    SiteBCalling_PT

    Create your route Pattern

    9.@ in SiteACalling_PT w/ whatever.RF to SiteA_RL

    9.@ in SiteBCalling_PT w/ whatever.RF to SiteB_RL

    Does this makes sense?

    Please rate post if this helps. Thanks.

  • RELEVANCY SCORE 2.95

    DB:2.95:Pri Utilization 1k



    I have a total of three PRIs connected to my Voice Gateways.  Two of them are listed as MGCP and one is H.323.  As we aren't doing any video that I am aware of, I question the need for this PRI.  Are there any handy commands that show the utilization of a PRI?  Is there any other reason to need a H.323 PRI?

    DB:2.95:Pri Utilization 1k


    You can use the Gateway Utilization Report that are generated under "CDR Analysis and Reporting" for PRI utilization.

    Suresh

  • RELEVANCY SCORE 2.93

    DB:2.93:Voice Gateways, Slt And Pgw jf



    Hi everyone,

    I dont know if its the right place to start this discussion so forgive me if i am wrong.( I also have opened a discussion in IP telephony portion)

    The company in which i am currently wokring is running a call center and IPT as well. i will try to explain the scenario

    voicegateways (AS5400 and AS5350 routers with IOS Version 12.4(15)T7)

    SLT (2651XM with IOS Version 12.2(8)T10)

    PGW (PGW 2200 with SunOS 5.10, MGC)

    I was under the impression that if you want to connect  PSTN and a voip network you need a PSTN gateway. So why are we using 3  different types of hardware.

    Follwing are the explainations in a doucment given to me

    the  Cisco PGW 2200 provides service providers with the capability to  seamlessly route voice and data calls between the PSTN and New  World  packet networks.

    Cisco 2611 Signaling Link Terminals (E1 terminates, Take Signaling part and send B n D channels to Voice gateway, Signaling part is resolved by it self and PGW)

    Voice Gateways

    allows  terminals of one type, such as H.323, to communicate with terminals of  another type, such as a PBX, by converting protocols. Gateways connect  an organization’s network to the PSTN

    Any information is much appreciated

    DB:2.93:Voice Gateways, Slt And Pgw jf


    i think in the link last date of support is July 2017 !

  • RELEVANCY SCORE 2.93

    DB:2.93:Balance Calls z8



    Hi,

    I have two h.323 gateway to the same route pattern, how can i do to balance the calls?, i tried with route group and route list, but always use the first gateway

    Regards

    DB:2.93:Balance Calls z8


    I think this possibility, i´ll try to do.

  • RELEVANCY SCORE 2.93

    DB:2.93:Gatekeeper 9x



    I have the following VoIP Scenario:

    Remote AS5300s or any other H.323 Gateways are connected through Internet to my AS5300 (equipped with two vfc cards and a quad card E1) connected to PSTN and calls are originated from remote H323 gateways to my AS5300.

    I want to add a gatekeeper to my network , so I prepare a 7200 VXR series route equipped with eight ethernet port card(available in my stock) to be the gatekeeper , this router will be placed near my AS5300 gateway.

    First question:

    I am asking if it is possible for the local AS5300( which is my AS5300) and the remote H323 gateways (spread around the Internet) to be all registered with the gatekeeper in one local zone ,in other way is it possible to create one local zone on the gatekeeper and put all my gateways ( local and remote) in this zone and then have my local AS5300 gateway registered with the gatekeeper using different tech-prefic to distinguish the remote gateways ?

    Second question:

    Could I use the 7200 VXR series as a gatekeeper and an ip gateway( between my local network and Internet ) at the same time ?

    please reply on this e-mail : jacob@teltac.com

    DB:2.93:Gatekeeper 9x


    I am sorry i couldn't post it yesterday. Please find it in Open Voice Application with the same heading. Thanks

  • RELEVANCY SCORE 2.92

    DB:2.92:H.323 Secondary Gateway Issue When E1 Service Fails x8



    Hi,

    We've had a problem today with one of our remote sites. The E1 service failed and the users were unable to receive inbound external calls (as expected) or to make outbound external call (not expected).

    The Call Manager 4.0 cluster has two H.323 gateways for external calls for each remote site (the primary is routed via the remote site E1 service and the secondary via the CCM site).

    The site has been configured so that if all ISDN ports are busy new external calls overflow to the secondary gateway. This has been tested and works fine. SRST has also been configured on the router and tests OK.

    The problem that happened today appears to be that when the E1 service failed the calls were still being routed to the primary H.323 gateway and not being sent to the secondary gateway. This was becaus the loopback address that binds the gateway address was still up up.

    Is there any way to configure the router or the call manager so that the above situation does not happen and that if the E1 service fails new calls are routed to the secondary gateway.

    Some configs below.

    Loopback config for H.323 gateway

    interface Loopback0

    description Loopback Address

    ip address 10.x.x.x 255.255.255.255

    h323-gateway voip bind srcaddr 10.x.x.x

    sh ip int br

    Serial0/0/0:15 unassigned YES NVRAM down down (E1 service down)

    Loopback0 10.x.x.x YES NVRAM up up (Loopback interface up)

    Route/Hunt list Configuration (in order of priority)

    Remote_RG[non-QSIG] (Remote site)

    Local_RG[non-QSIG] (CCM site)

    Note: When the route list priorities were changed manually the remote site could make external outbound calls via the secondary gateway.

    Thanks in advance for help provided.

    Regards,

    Bryce.

    DB:2.92:H.323 Secondary Gateway Issue When E1 Service Fails x8


    We think we've found the answer to the issue however we haven't been able to test it yet.

    On the Call Manager in the Service Paramater Configuration page you chose true for Automated Alternate Routing Enable*

    Will advise how it went. Thanks for help.

  • RELEVANCY SCORE 2.92

    DB:2.92:Call Redirection In H.323 Gateways jx



    Hi, we have 2 voice gateways. In GW1 3 tata links has been terminated in GW2 3 Airtel links are terminated.according our requirement all local calls will go through airtel links i.e GW2 and all ISD call will go through Tata Link i.e GW1. for local calls, in GW2 all the airtel links are busy the call has to be routed to GW1 tata links.the same way GW1 all tata links are busy ISD calls has to go through GW2 airtel links. so how can we achieve this. if this is possible please let me know the logiic....

    DB:2.92:Call Redirection In H.323 Gateways jx


    Hi there,

    I assume you have Call Manager there, so use Route Groups and Route List in your patterns.

    Check for more information Cisco Unified Communications Manager Administration Guide: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_1_1/ccmcfg/b03rtgrp.html

  • RELEVANCY SCORE 2.92

    DB:2.92:How Many Gateway Can 3600 Router Support As A Gatekeeper sj



    I like to check how many gateways a Cisco 3640 can support as a H.323 gatekeeper. Is there a difference in the number for Cisco (AS5300) and Non-Cisco gateways?

    DB:2.92:How Many Gateway Can 3600 Router Support As A Gatekeeper sj


    Cisco is recommending 100 registered endpoints per gk ( no difference between endpoints),

    it's based upon a desired call/second rate of 80 cps. The basic concept is that as you increase the number of supported endpoints, the performance

    in calls/second will decrease.

  • RELEVANCY SCORE 2.92

    DB:2.92:Rtmt Call Active dk



    Hi

    My RTMT show me 4 active calls, but on voice gateway (show voice call status) show me 14 active calls.

    My gateways is H.323 and is a link ISDN PRI. Is there any configuration on RTMT/CCM/Gateway?

    Thanks

    Peterson

    DB:2.92:Rtmt Call Active dk


    Yes.

    I suspect that the RTMT shows just outbound calls......

    Tks

    Peterson

  • RELEVANCY SCORE 2.92

    DB:2.92:Gateway Migration From H.323 To Sip 97



     

    Hi All,

    Currently all my Cisco Gateways registered with CUCM using H.323 protocol, for now i would like to migrate all these gateways registered protocol to SIP, simply i would like to Migrate all my H.323 gateways to SIP.

    Can anyone have any best procedure to do this Migration, as well share the related documents if you have any.

    Thanks in Advance for your Help.

     

    Regards,

    Madhu

     

    DB:2.92:Gateway Migration From H.323 To Sip 97


    Madhu,

    There are a few things you need to consider..

    1. You need to create sip trunks on cuc and point the trunks to the ip address of your gateway. If you have multiple IP address on the gateway, you need to carefully consider which ip address you will bind your sip signalling to.

    2. You need to ensure that the cucm group in the device pool you assign to your sip trunk is the same as youconfigure for your dial-peers or you use the feature "run on all active cucm" on your sip trunk and RL.

    3. Configure inbound dial-peer from cucm to the gateway and ensure it is enabled for sip (by using sip protocol sipv2 command)

    4. Configure outbound dial-peer to cucm and ensure it is sip enabled.

    5. configure rtp-nte on both your inbound and outbound dial-peers fir dtmf

    6. ensure your sip trunk is set to use "no preference for DTMF

    To understand more on sip signalling please refer to this document

    https://supportforums.cisco.com/blog/154506

     

  • RELEVANCY SCORE 2.91

    DB:2.91:Monitor T1/E1 Pri Utilization m9



    Does anyone know of any software, Cisco or 3rd party that can monitor real time PRI utilization? Also, can this sends alerts and reports when it reaches a predetermined threshold?

    We have CCM 4.1.3 and are using H.323 gateways.

    DB:2.91:Monitor T1/E1 Pri Utilization m9


    Cisco Unified Service Statistics Manager will be a better option for monitoring real-time PRI utilizaton.Cisco Unified Service Statistics Manager (Service Statistics Manager) is a product from the Cisco Unified Communications Management Suite that uses short-term operational data-collected by other products in the suite-to perform longer-term analysis and reporting.

    Refer the "User Guide for Cisco Unified Service Statistics Manager" present in the url below:

    http://www.cisco.com/en/US/docs/net_mgmt/cisco_unified_service_statistics_manager/1.0/user/guide/ssm_ug.pdf

  • RELEVANCY SCORE 2.91

    DB:2.91:The Cisco Callmanager Find And List Page Status Field For Gateways Is Shown As Unknown 77


    Core Issue
    When the gateways are added, it is shown as UNKNOWN in the status field in the Cisco CallManager Find and List Gateways page. However, the calls are routed across the gateways properly.

    Resolution
    This is normal behavior when the gateway is an H.323 gateway as it is a peer-to-peer relationship. However, for an Media Gateway Control Protocol (MGCP) gateway, it needs to be registered with the Cisco CallManager as it is a master/slave setup.

    DB:2.91:The Cisco Callmanager Find And List Page Status Field For Gateways Is Shown As Unknown 77

    Core Issue
    When the gateways are added, it is shown as UNKNOWN in the status field in the Cisco CallManager Find and List Gateways page. However, the calls are routed across the gateways properly.

    Resolution
    This is normal behavior when the gateway is an H.323 gateway as it is a peer-to-peer relationship. However, for an Media Gateway Control Protocol (MGCP) gateway, it needs to be registered with the Cisco CallManager as it is a master/slave setup.

  • RELEVANCY SCORE 2.91

    DB:2.91:Urgent!!!! Voice Gateway Was Hacked, Were Made Thousand Of L.D Calls kf



    I have several 2800 Voice Gateways in several regions. How can I protect my H.323 GW? these Gateways have public IP addresses. Can I control or Authenticate my VOIP Gateways in order to eliminate a rogue Gateway can connect to my Gateway and they can make calls?

    DB:2.91:Urgent!!!! Voice Gateway Was Hacked, Were Made Thousand Of L.D Calls kf


    I think this needs some clarification, because it can be a very important issue.

    First, I would check out this tip:

    http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_tech_note09186a00809dc487.shtml

    Then, you need to understand these basics:

    H323 will respond to any request on any interface on TCP port 1720 by default. This applies even if you have the bind command.

    SIP will respond on any IP address on UDP/TCP 5060 by default as long as the router has a voice-port. If you bind the media address, SIP will only respond on that address.

    This means in short - even if you're running an H323 only gateway, you are still capable of bouncing incoming SIP traffic out your dial peers. It is very common to see a H323 only gateway with a .T or 9.T pots dial peer, and attackers hit the public IP address, it matches an incoming voip dial peer (or dial-peer 0), and then the wildcard matches the PSTN pots wildcard dial peer.

    If you have a public IP address, make sure that you disable all SIP traffic on TCP/UDP ports 5060. You can use 'show tcp' or 'show ip socket' or 'show udp' to see some of the open ports.

    HTH,

    Nick

  • RELEVANCY SCORE 2.91

    DB:2.91:Ccm Load Balancing sm



    I have a 5300 E1 ports on two different gateways used for incoming. I have configured the route patterns in the callmanager poining to one gateway only. I have configured all of them with h.323. I have to load balance the outgoing call on both the gateways. The site is non NANP. How can I achieve this.

    DB:2.91:Ccm Load Balancing sm


    I just checked my 3.3(3) system, and it says flag. So if it did say that at one point, its fixed.

  • RELEVANCY SCORE 2.90

    DB:2.90:H.323 Gw Failure m7



    have route list with 3 h.323 gw's.

    congle 323 gw in each RG.

    I pulled the PRI out of first 323 gw and made test call and the call did not go through.

    i have the route pattern pointed to the route list with the 3 h.323 gw's but does not seem to be hitting next member in RL.

    is there a special config setting in service params that is not a default settin gthat i must change for this to work or could this be an issue on my 323 GW?

    DB:2.90:H.323 Gw Failure m7


    From a previously question on the same answered by gogasca:

    H323 Failover

    The default behavior in IOS, when there are no valid POTS dial-peers for a call to go out of, is to return “unallocated number” as a cause code to CCM. This can happen even if a t1 goes down. When the T1 goes down, the POTS peer is marked as down, and when CCM sends a call to the gateway, it'll return a UAN cause code to the ccm, causing CCM (by default) to stop hunting for other available gateways. There are ways to change the behavior using service parameters in CCM, but this behavior didn't make sense to me, so we looked for other ways to do it.

    It turns out that you can issue the global command "no dial-peer outbound status-check pots" on the IOS GW, if you're opposed to changing the CCM behavior. What this command will do is cause the dial-peer to stay up. IOS will try and route the call, and when the T1 is down, it returns "No circuit available" to CCM. When CCM receives this cause code, it knows there's been a non-user error, and continues hunting, achieving the desired behavior.

    In your case since it is H323 it will be retruning only H225 release comp message with proper code to continue re-routing.

    The IOS code will translate any code we get from pots side to IP side

    HTH

    javalenc

    if this helps, please rate

  • RELEVANCY SCORE 2.90

    DB:2.90:Fax Email And Mgcp Solution? cs



    Is there the solution for faxes to forward them to email with MGCP gateways? Or I should use only H.323?

    DB:2.90:Fax Email And Mgcp Solution? cs


    T.37 Store-and-Forward fax is not supported on voice ports controlled with MGCP. Other voice ports on this gateway that are not controlled by MGCP can be used for T.37. H.323 and SIP do not have this same restriction so one of these call control protocols is preferred when T.37 functionality is needed.

    Regards,

    David

  • RELEVANCY SCORE 2.89

    DB:2.89:User Finds It Difficult To Configure Fax Relay Between A Cisco 6624 Skinny Or Mgcp Gateway And An H.323 Cisco Ios Gateway m3


    Core Issue
    When configuring fax relay between a 6624 gateway and an Cisco IOS  H.323 gateway, problems may arise due to a mismatch between the two fax modes the gateways are running. In order for the fax to be successful, the two gateways must negotiate the same fax mode. By default, a Cisco router and gateway uses fax-relay negotiation. However, as from the 3.010 load for WS-X6624 (A002A3A0), the default fax mode is fax pass-through.

    Resolution
    Ensure that the fax mode matches between the gateways. In particular, if you leave the CallRestartTimer with the default setting of 5000 ms, the 6624 port uses fax pass-through. To use fax relay, change this value to 1235. The other option is to use fax pass-through on the H323 gateway.
    For the Cisco CallManager configuration, perform these steps:

    Open up the Cisco CallManager Administration panel.Click the 24 port Foreign Exchange Station (FXS) blade configured on your Cisco CallManager.Select the port you are using (port to which your fax has been connected).Set the CallRestartTimer to 1235 ms. Click Update.
            Note: This procedure only applies to skinny gateways, not to H.323 gateways.
    If you leave the CallRestartTimer with the default setting of 5000 ms, the port uses fax pass-through. If that is the case, make sure to also use fax pass-through on the H.323 gateway.

    To configure the H.323 gateway fax relay, set the fax-rate command to 9600 or 14400 baud in cases where the default G.729 codec is being used, as shown. When using G.711, this command is not required.

    dial-peer voice 2 voip
    destination-pattern 2000
    session target ipv4:10.200.72.37
      fax-relay ecm disable
      fax-rate 9600
    !

    dial-peer voice 3 pots
    destination-pattern 0T
    direct-inward-dial
    port 1/0:15

    To configure the H.323 gateway for fax pass-through, issue the fax rate disable command on the voip dial peer, as shown:

    dial-peer vo
    ice 2 voip
    destination-pattern 2000
    session target ipv4:10.200.72.37
      fax-relay ecm disable
      fax rate disable
      no vad
    !
    dial-peer voice 3 pots
    destination-pattern 0T
    direct-inward-dial
    port 1/0:15

    For more information about configuring faxes, refer to Fax configuration on a Cisco WS-X6624 Using an H.323 Gateway.
    Voice Gateways
    IOS gateways

    Non-IOS gateways

    DB:2.89:User Finds It Difficult To Configure Fax Relay Between A Cisco 6624 Skinny Or Mgcp Gateway And An H.323 Cisco Ios Gateway m3

    Core Issue
    When configuring fax relay between a 6624 gateway and an Cisco IOS  H.323 gateway, problems may arise due to a mismatch between the two fax modes the gateways are running. In order for the fax to be successful, the two gateways must negotiate the same fax mode. By default, a Cisco router and gateway uses fax-relay negotiation. However, as from the 3.010 load for WS-X6624 (A002A3A0), the default fax mode is fax pass-through.

    Resolution
    Ensure that the fax mode matches between the gateways. In particular, if you leave the CallRestartTimer with the default setting of 5000 ms, the 6624 port uses fax pass-through. To use fax relay, change this value to 1235. The other option is to use fax pass-through on the H323 gateway.
    For the Cisco CallManager configuration, perform these steps:

    Open up the Cisco CallManager Administration panel.Click the 24 port Foreign Exchange Station (FXS) blade configured on your Cisco CallManager.Select the port you are using (port to which your fax has been connected).Set the CallRestartTimer to 1235 ms. Click Update.
            Note: This procedure only applies to skinny gateways, not to H.323 gateways.
    If you leave the CallRestartTimer with the default setting of 5000 ms, the port uses fax pass-through. If that is the case, make sure to also use fax pass-through on the H.323 gateway.

    To configure the H.323 gateway fax relay, set the fax-rate command to 9600 or 14400 baud in cases where the default G.729 codec is being used, as shown. When using G.711, this command is not required.

    dial-peer voice 2 voip
    destination-pattern 2000
    session target ipv4:10.200.72.37
      fax-relay ecm disable
      fax-rate 9600
    !

    dial-peer voice 3 pots
    destination-pattern 0T
    direct-inward-dial
    port 1/0:15

    To configure the H.323 gateway for fax pass-through, issue the fax rate disable command on the voip dial peer, as shown:

    dial-peer vo
    ice 2 voip
    destination-pattern 2000
    session target ipv4:10.200.72.37
      fax-relay ecm disable
      fax rate disable
      no vad
    !
    dial-peer voice 3 pots
    destination-pattern 0T
    direct-inward-dial
    port 1/0:15

    For more information about configuring faxes, refer to Fax configuration on a Cisco WS-X6624 Using an H.323 Gateway.
    Voice Gateways
    IOS gateways

    Non-IOS gateways

  • RELEVANCY SCORE 2.89

    DB:2.89:In Band Signaling With Sip Tdm Gateway And Ccm 9c



    I am starting to prefer using SIP instead of H.323 for TDM gateways in the newer versions of call manager because of SIP having better support for globalization (doesn’t drop the “+” in VoIP peers). However I am running into an issue with in-band signaling messages from telco. If I route a call and it fails (number out of service), the IP phone caller hears endless ringing until the call times out at which point it goes to fast busy. Changing the SIP profile in CCM to enable early offer appears to resolve this problem, but I have also seen some references to using ‘progress_ind alert enable 8’ on the POTS dial peer too which was something I used to do when troubleshooting no ring back in H.323 gateways in the past. I am wondering if one method would be superior to the other. The early offer method also seems to resolve issues with connections to some IVR’s that start playing their initial greeting before a Q.931 connect message has been sent. In the H.323 world, I used to use ‘voice rtp send-recv’ to resolve that which would also sometimes happen with 911 calls (PSAP would never send a connect). I know this is kind of a rambling thread, but I am trying to create a better recipe going forward and trying to determine the comparative advantages of one method over the other.

    DB:2.89:In Band Signaling With Sip Tdm Gateway And Ccm 9c


    Dear

    I need to help you , Kindly be informed that the SIP uses in-band where as the SCCP uses out-of band .To solve this problem you have to use MTP .You will enable the MRGL in the SIP trunk.

    Thank you

    Please rate if this will help

  • RELEVANCY SCORE 2.89

    DB:2.89:+ In Calling Number H.323 7x



    Hi All,

    I've setup H.323 GW in CUCM to point to SP. we need to send full E.164 number for calling party "+19495551234". I've added "+" in route pattern calling party , also created calling party transformation to add "+" in calling party and assign to H.323 gateway, but still SP sees calling number without "+".

    any suggestion would be very appreciated.

    thanks

    Alex

    DB:2.89:+ In Calling Number H.323 7x


    H323 gateway afaik doesnt support "+". You will have to either use MGCP or SIP.

    Please rate useful posts.

  • RELEVANCY SCORE 2.89

    DB:2.89:Mgcp Gateways And Cisco Fax 9m



    We are considering adding a fax solution to our IPT infrastructure using Cisco Fax (Captaris RightFax). Ideally we would like the same voice gateways to terminate the fax calls (internal voice calls use QSIG and external calls use Q.931). Can this be achieved with MGCP-controlled Voice gateways? Can T.38 fax be configured with an MGCP gateway. I have seen examples where H.323 dial-peers are required to talk SIP to the Fax Server.

    If not, is an alternative to use a back-to-back PRI connection between a voice gateway port and the Fax Server ISDN card, with appropriate route patterns in CallManager to route the fax calls to Fax Server, using Cisco Fax Relay?

    DB:2.89:Mgcp Gateways And Cisco Fax 9m


    Hey Guy,

    Just curious how you got t38 to work with MGCP? I work with RF a couple times a month and I have not been able to get that to scream yet?

    Thanks

  • RELEVANCY SCORE 2.88

    DB:2.88:Delayed Voice For Attendant Console - H.323 Gateways f3



    I have had a customer with attendant console mention that it takes a little while for the voice to be heard when answering the call. CCM 3.3(2)SPC and using 2651XM gateways running H.323, ISDN PRIs. I was looking at a few other conversations on this forum and saw a mention of some H.323 timers that could possibly be tuned.

    The H.323 faststart inbound option. Would this help? I am reluctant to change things like this with the amount of users on the system as other than this the network is AOK.

    Any suggestions?

    DB:2.88:Delayed Voice For Attendant Console - H.323 Gateways f3


    Thanks Paul... just wanted to make sure there were no adverse affects if I put it on. And voice ctp send-recv is definitely on.

    Cheers

  • RELEVANCY SCORE 2.87

    DB:2.87:Gatekeeper With A Gateway Farm k9



    Hi, do you now if i can do a parcial migration of 2 gateways as gatekeepers endpoints leaving the other gateways as part of the normal voip network. I am asking that because the customer have 10 gateways that conform the h.323 network and he want to test the same network but using a gatekeeper, so if i put only 2 of the gateways as gatekeepers endpoints, these will affect the normal call processing or not?

    Thanks in advanced

    Jose

    DB:2.87:Gatekeeper With A Gateway Farm k9


    Jose,

    This is certainly feasible. Just be sure that you have removed the references to these 2 gateways in route lists, route patterns, etc. and be sure that you have route lists and route patterns in place that point to the new gatekeeper.

    Hope this helps. If so, please rate the post.

    Brandon

  • RELEVANCY SCORE 2.87

    DB:2.87:Cisco Call Manager 5.1 And Cisco Voip Gateways 9c



    Hi,

    I have Cisco Call Manager 5.1 with Cisco 2800 series routers as a ISDN gateways and Cisco VG224/ATA 186 as POTS gateways. While Cisco Call Manager collects call statistics from 7900 series SCCP handsets managed by the Call Manager (number of VoIP packets, dropped frames etc.), it shows only one side of the call as it does not collect statistics from the Cisco routers (configured as H.323 gateways), the Cisco VG224 (with ports configured as H.323 gateways and as a SCCP devices) and Cisco ATA 186 (configured for SCCP).

    Is the collection of these statistics possible? What needs to be done?

    Thanks,

    Paul

    DB:2.87:Cisco Call Manager 5.1 And Cisco Voip Gateways 9c


    Update: Logging of QoS/Call Management Records (CMR) to Call Manager CDR database is not possible with H.323 gateways; only MGCP gateways:

    http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a008020650a.shtml#qa4

    Anyone know how to get the Cisco router gateway to log these statistics anyway using standard router logging mechanisms (e.g. syslog)?

    Anyone know about calls to SCCP ports on Cisco VG224 or ATA 186?

    Thanks,

    Paul

  • RELEVANCY SCORE 2.87

    DB:2.87:Mgcp H.323 Gateways - Pros Cons? s9



    I'm just curious about what everyone thinks about the Pros Cons of MGCP and h.323 Gateways with CCM. I know MGCP Gateways are controlled by CCM and h.323 are all router controlled, but what about stuff like: Are calls dropped when the gateways are reset or will the gateway not reset till all calls are over? Can you run gateway utilization reports on both types of gateways? Any pros and cons would be appreciated! Thanks!

    DB:2.87:Mgcp H.323 Gateways - Pros Cons? s9


    a comparison of h.323 and mgcp is documented here:

    http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00806fedbe.shtml

    I think the discusssion about pro contr. of the protocols is obsolete, because the choice of protocol depends from the requirements of the customer.

    An example: if you need QSIG Integration, you have the best interoperability with MGCP.

    regards

    mehmet

  • RELEVANCY SCORE 2.87

    DB:2.87:How Many Gateway Can 3600 Router Support As A Gatekeeper z8



    I like to check how many gateways a Cisco 3640 can support as a H.323 gatekeeper. Is there a difference in the number for Cisco (AS5300) and Non-Cisco gateways?

    DB:2.87:How Many Gateway Can 3600 Router Support As A Gatekeeper z8


    Cisco is recommending 100 registered endpoints per gk ( no difference between endpoints),

    it's based upon a desired call/second rate of 80 cps. The basic concept is that as you increase the number of supported endpoints, the performance

    in calls/second will decrease.

  • RELEVANCY SCORE 2.87

    DB:2.87:Cucm 7.1.3 Outbound Calls To Multiple Gateways Using Route List / Route Group jm



    Hi all,

    I am running CUCM 7.1.3 and I am using Route List and Route Group to route outbound calls to multiple gateways in case if one gateway fails, calls will get routed to the other gateway. But for some reason it didn't seem to work for me today where two of the PRI's on one of the gateways fail and outbound calls did not failover to the other gateway in the Route List. Has anyone seen this issue ?

    In this Route List I have two Route Groups and each Route Group contains a gateway and the gateways are H.323 gateways. My thinking is that if the PRI circuits are down on one gateway calls will get routed to the next gateway in the Route List. Is that correct ?

    Thanks all in advance !! I appreciate any inputs / suggestions !!!

    Danny

     

  • RELEVANCY SCORE 2.87

    DB:2.87:Redundant Gateway Configurations To Provide High Availability az



    Could anyone provide me with configuration for redundant voice gateways in a branch office, H.323 and MGCP?

    Thank you

    Marco

    DB:2.87:Redundant Gateway Configurations To Provide High Availability az


    This URL should help you:

    http://www.cisco.com/en/US/products/ps6600/products_data_sheet09186a00800b4694.html