• RELEVANCY SCORE 3.79

    DB:3.79:Problems Routing Calls Between Cme And Ccm ds





    Iam having issues routing calls from a phone registered to CME to a CCM phone

    Calls in the CCM-CME direction works perfectly fine

    On the CCM the following things have been done

    Configured the CME as H323 gateway

    Created non gatekeeper controlled ICT trunk

    Created a route pattern for the destination and pointed to the above gateway

    On the CME

    configured a dial-peer with a destination pattern and session target pointing to CCM

    Any Ideas what could be the problem?

    Narayan

    DB:3.79:Problems Routing Calls Between Cme And Ccm ds


    There are no partitions configured on the system

    The RP for eg 15xxx is configured to point to CME (configured as H323 gateway)

    BTW i got to make it work somehow

    Narayan

  • RELEVANCY SCORE 3.63

    DB:3.63:Intercluster Cme Calls Using 7985? m7





    Hi there,

    I need to verify. Can we connect two CME systems to make video calls using 7985 SCCP phones? or can we make video calls using 7985 between a CCM and CME as well?

    DB:3.63:Intercluster Cme Calls Using 7985? m7


    Yes, assuming IP connectivity and sufficient bandwidth.

    Brandon

  • RELEVANCY SCORE 3.59

    DB:3.59:Cme Or Ccm Integration In To A Mitel 3300 3z





    Does anyone know if it’s possible to get basic call functionality working between CME or CCM and a Mitel 3300 using a H.323 link, or is this only possible over a PRI link?

    Any experience on this would be appreciated.

    DB:3.59:Cme Or Ccm Integration In To A Mitel 3300 3z


    Does anyone know if it’s possible to get basic call functionality working between CME or CCM and a Mitel 3300 using a H.323 link, or is this only possible over a PRI link?

    Any experience on this would be appreciated.

  • RELEVANCY SCORE 3.52

    DB:3.52:Trunk Type For Cme And Ccm da



    What type of trunk does Cisco recommend between CME and CCM, H225 or intercluster? Thanks

    DB:3.52:Trunk Type For Cme And Ccm da


    You will use an ICT to integrate CCM and CME. Please see the following link for the configuration steps necessary to make this work.

    http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00805b21fd.html#wp1057434

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

    Brandon

  • RELEVANCY SCORE 3.52

    DB:3.52:Sip Trunk Between Cme And Cm 6.1 ka



    Hi All,

    I was hoping someone could point me in the right direction on how to create inter-cluster trunks? I'm assuming this would be the best was to make VOIP calls between my CME site and all other sites registered to CM?

    I know i can convert the CME site to an H323 gateway and get the phones to register to CM, which is what I've done to other remote sites. But i have to specifically keep this site on CME.

    i know i can create multiple voip dail-peers with the particular extension range for remote sites and point them to CM, but i'm thinking a SIP trunk between CME and CM would be better.

    Any doco with config examples for CME and CM setup would be greatly appreciated.

    Cheers

    DB:3.52:Sip Trunk Between Cme And Cm 6.1 ka


    Thanks for your quick reply Java.

    I've created a SIP trunk on CM pointing to the CME (10.1.144.1) and I've applied a CSS which contains all the remote site partitions which are registered to CM.

    I've also added what i believe is the correct config on the CME to allow SIP trunk calls, but I might be missing something.

    Attached are screen shoots of my SIP trunk and CME.config added, not sure if i'm missing something small.

    I'd appreciate any help you could provide.

    PS, The customer has specifically requested that the site remain as CME as the end users are quite particular. I've already convereted several CME sites to h323 gateways.

  • RELEVANCY SCORE 3.49

    DB:3.49:Cme Blocking Calls Between Extensions s7



    Hi all,

    I am trying to configure UC560 to block internal calls to certain extensions, so to speak we have pool of extensions 2040 to 2070 and only one extension 2010 is allowed to place calls to an extension 2069, any other extension must not be able to dial extension 2069. Is it possible to accomplish it with CME? I know you can do it quite easily in CallManager by configuring and manipulating Call Search Spaces but not sure if you can do it with CME or UC560.

    Thanks for any suggestions.

    DB:3.49:Cme Blocking Calls Between Extensions s7


    Hi Remi,

    The dial-peers which you have configured will not be hit when dialling between internal numbers.Hence i would suggest you to try the following:-

    Since you want to block all numbers between 2040 and 2070 from dialling 2069, create a translation rule and profile as follows:-

    voice translation-rule 1 rule 1 reject /20[4-7]0/

    voice translation-profile REJECT translate calling 1

    Then go to the ephone-dn (which has number 2069) which is automatically created when IP phones register to CME. You can use the command 'show ephone-dn' to help yourself out. (Lets assume for ext 2069 your ephone-dn is 50)

    ephone-dn  50     number 2069     translation-profile incoming REJECTIf the above doesn't work(depending on your IOS) try this:- ephone-dn  50     number 2069     translate calling 1

    HTH

    Regards

    Nitesh

    PS: Please rate helpful posts

  • RELEVANCY SCORE 3.43

    DB:3.43:Ccm/Cme And Cue Questions df



    Hi,

    CCM/CME - An employee places a call into the IP PBX, enters the destination number they are trying to call. The PBX then disconnects the caller, places the call to the destination, calls back the first caller and then connects the call between parties. Customer says Avaya PBX can do this, does Cisco have a solution for this?

    CUE - Is it possible to customize the voice message prompt for individual voice mail box. User wants their greeting to say “Press 1 for Cell, Press 2 for home etc. Why they cannot just provide this information on the standard greeting is beyond me. Still it will be nice to know if I can do this with CUE

    DB:3.43:Ccm/Cme And Cue Questions df


    Thanks

    CUE - Yes, i know how to do this with Unity, i will have to get creative with a cue script that can do this.

  • RELEVANCY SCORE 3.41

    DB:3.41:Routing Calls From Cme To Ccm Through A Gatekeeper Testing With Csim Start 73



    I am trying to route calls from cme to ccm through a gatekeeper. Everything seems to be working fine except when I test the calls from the router using csim start I cannot get the IP phone on the ccm side to ring. I have tech prefix set to 1# on ccm h323 trunk and 2# on cme. I have tech prefix set on and session target ras on dial-peer pointing to the gatekeeper. I have a translation pattern set in ccm to strip the 1# off of the called number from cme. when I do a show h323 gateway I see the no response from user count increment everytime I make a csim start test call. When I do a debug gatekeeper main 10 I can see where the tech prefix is matched and the zone is matched. Nothing happens after that. By the way calls from ccm to cme work. Just not the other way around.

    DB:3.41:Routing Calls From Cme To Ccm Through A Gatekeeper Testing With Csim Start 73


    If you must use a tech-prefix for each site, then you must enter the CCM address explicitly.

    gw-type-prefix 1#* gw ipaddr {address of CCM}

    Hope this helps.

  • RELEVANCY SCORE 3.40

    DB:3.40:Cme 4.2 Srst With H323 Ccm 4.1.3 sj



    Hi,

    We currently have a local CCM 4.1.3 cluster with local E1 gateways and need to add a new remote site.

    The solution I'm building (in lab first) will use a CME 4.2 on the remote site configured with SRST in case the link falls.

    When the link is up, the remote users should register to the CCM but still use the CME as their gateway out to the PSTN.

    For this to work, I have configured the CME as a h323 gateway in CCM.

    For outbound calls all is working fine, for inbound calls, it seems that the CME is matching it's own dial-peers first.

    When the link is up this means that the phones will not receive inbound calls.

    To fix this issue, I have added the $ in my voip dial-peer destination patterns.

    When doing this inbound calls work as well, but the problem is that the phones only start ringing after the caller has allready heard about 2 or 3 rings on his phone.

    Is there a way to get this time down so that the phones start ringing directly?

    Thanks in advance fort the help.

    Jeroen.

    DB:3.40:Cme 4.2 Srst With H323 Ccm 4.1.3 sj


    Hi,

    Thanks for your answer.

    Making the gateway MGCp is not an option as we need to have support for T38 faxing using a fax over Ip solution. With our version of callmanager, this only works with h323.

    This is a piece of my config, you will see that it is configured with the srst mode auto-provision none command :

    telephony-service

    srst mode auto-provision none

    srst ephone description Cisco Unified CME SRST Fallback : Aug 17 2007 14:53:32

    srst dn line-mode dual

    fxo hook-flash

    load 7960-7940 P00308000400

    max-ephones 24

    max-dn 72

    ip source-address 10.10.1.77 port 2000

    calling-number initiator

    timeouts interdigit 3

    system message SRST active.Limited Features

    time-zone 5

    time-format 24

    date-format dd-mm-yy

    max-conferences 8 gain -6

    call-forward pattern .T

    call-forward system redirecting-expanded

    moh music-on-hold.au

    web admin system name admin password engineer

    dn-webedit

    time-webedit

    transfer-system full-consult dss

    transfer-pattern 0.T

    secondary-dialtone 0

    create cnf-files version-stamp 7960 Aug 20 2007 11:30:00

    Thanks,

    Jeroen

  • RELEVANCY SCORE 3.37

    DB:3.37:Gatekeeper H.225 Trunk Between Ccm And Cme dd



    CCM4.13 with Unity 4.21 and CME 3.3 with CUE 2.34

    Everything is working fine, except some little stuff.

    For example, the voice mail pilot 1000 at CME, voice mail pilot 2000 at CCM.

    At CME side, I dial 2000 to check my voice mail at Unity, which is fine.

    However, if I dial 1000 at CCM phone, I cant get to CUE to check voice mail, which is same as AvT (1001).

    I believe pilot 1000 or AvT (1001) are not part of extension. Is there anyway to let me able to check CUE voice mail from CCM side by dialing pilot 1000 or AvT 1001?

    Thanks

    Ken

    DB:3.37:Gatekeeper H.225 Trunk Between Ccm And Cme dd


    Shanky

    I know what is wrong my problem now.

    Since I do NOT define zone prefix for CME side, therefore the DID and extension is automatically register to gatekeeper when I do "show gateke end".

    Therefore, I have to do "dialplan-pattern 1 21232110.. extension-length 4 no-reg" and "number 1001 no-reg both" under ephone-dn.

    Then I add "zone prefix cme 10.." at the gatekeeper. Then I can dial the pilot number 1000 now.

    Since the pilot number 1000 wasnt register(not part of extension) to gatekeeper before which is normal.

    Is that make sense to you???

    Thanks

    Ken

  • RELEVANCY SCORE 3.31

    DB:3.31:Ccm Intercluster Trunk To Cme 3.1 k3



    This is my first experience with CME, and I have not spent much time troubleshooting yet, but perhaps someone who has been down this road can save me some effort. I have CME3.1 running on a 3745 (12.3.7T)code, and my Call Manager is 3.3.3 SR1.

    No problem with calls between the CME phones, but G711 intercluster calls are dropped as soon as they are answered. The phone rings normally, and caller name is working. A debug H225/H245 seems to indicate a rejection of the capabilities exchange. I started with this set up for G729, but this gave me "not enough bandwidth" fast busy on the CCM originated calls, and "number not found" on the CME, so I went G711 with no location to see if it was a CAC issue.

    I have followed the doc on internetworking CME 3.1 with Call Manager, which is quite clear. There are only 2 Call Manager H323 parameters to set, and most of the critical config setting in the router are default. I did try setting them all just to be sure, but no change.

    Thanks, Dave

    DB:3.31:Ccm Intercluster Trunk To Cme 3.1 k3


    I got it working. Funny how things can be a lot clearer on a Monday morning than they were at 6:30 on Friday.

    The capabilities was being rejected because G729B (silence suppression) was disabled on the Call Manager, and that's what I had in the voice class on the router. The original insufficient bandwidth problem was because the Call Manager allocates 80kb per call even though it is a G729 region and the call is going through as G729. This is an apparent bug, but the simple workaround is to just set the location bandwidth for 240 which will allow the three G729 calls my QOS can handle. I tested this and it works.

  • RELEVANCY SCORE 3.24

    DB:3.24:Known In Outgoing Calls 9z



    Dear

    i have two cme (CME--- CME) between them trunk  , one of them version 12.4 and other is 15.1 , the problem is

    showing me " unknown " when i do outgoing calls .

    when i do debug

    i can see the  correct calling number and called number in both sides (CME)

    when i put CUCM instead of CME in the other side (CME----- CUCM) 

    everythings working fine without any troubles

    what do  you think  the problem ? and how to solve it ?

    DB:3.24:Known In Outgoing Calls 9z


    Dear

    i have two cme (CME--- CME) between them trunk  , one of them version 12.4 and other is 15.1 , the problem is

    showing me " unknown " when i do outgoing calls .

    when i do debug

    i can see the  correct calling number and called number in both sides (CME)

    when i put CUCM instead of CME in the other side (CME----- CUCM) 

    everythings working fine without any troubles

    what do  you think  the problem ? and how to solve it ?

  • RELEVANCY SCORE 3.20

    DB:3.20:Point-To-Point Video Calls Over Pstn/Isdn 98



    Hi,

    We have two independent locations,one of them has CUCM 6.1 and the other side, CME 7.1. Both can make video calls locally (with 7985s and CIPC).

    Is it possible to stablish point-to-point video calls between those two sides natively over PSTN/ISDN? Is it necessary another device - mcu - or something like that?

    I will appreciate your help because I am new in video management over CCM/CME environments

    Thank you in advanced

    DB:3.20:Point-To-Point Video Calls Over Pstn/Isdn 98


    Thank you both for your help, as I mentioned I am new in this area and need your advice. I will suggest IP network as you recommend, although I will also work on the document you offered because this is part of a solution that requires the use of ISDN and Polycom devices so, you will see some other posts from me , please forgive my ignorance and, again, thank you very much for your help.

  • RELEVANCY SCORE 3.18

    DB:3.18:Cme 3.1 / Ccm 3.2(2c) One-Way Calling kj



    In my lab, I'm trying to get a CCM 3.2(2c) at central site a CME in the branch to make calls.

    I can call CCM = CME but not from CME = CCM.

    In the branch I have a VoIP dial peer pointing to the Publisher for CCM DN's.

    A debug CCAPI shows the dial-peers matching but then the call get's torn down with a disconnect cause=0x26.

    Do I need to do anything else on the CCM to get incoming calls ?

    DB:3.18:Cme 3.1 / Ccm 3.2(2c) One-Way Calling kj


    Make sure the codecs are correct on both the sides.I have the same setup in the lab and it works fine.

    In the voip dial-peer you can add voice class codec command and in the global config create a codec class with different preference.

    Thanks,

    Radhika.

  • RELEVANCY SCORE 3.18

    DB:3.18:Ccme And Ccm Intercluster Trunk? 7c



    Anyone have any idea how an intercluster trunk can be setup between CCME (3.0) and a CCM cluster? I know I can use a gatekeeper to route calls between the two, but I'm trying to figure out if it can be done without a gatekeeper.

    DB:3.18:Ccme And Ccm Intercluster Trunk? 7c


    ICT is not supported with CME 3.0, but is supported on CME 3.1. The link below explains how to configure ICT in CME 3.1

    http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123newft/123t/123t_7/cme31sa/cme31bsc.htm#wp1001000

    If you have CME 3.0, on the CCM add the CME router as a H323 gateway. Use route pattern to route calls to CME. On CME use voip dial-peers to send calls to CCM.

    Hope this helps!

    Partha

  • RELEVANCY SCORE 3.18

    DB:3.18:H.323 Gateway Configuration Between Callmanager Express And Callmanager kf



    Scenario:

    The 7940 (SCCP) sets on the CME site can make and recieve calls to / from the CM site.

    The 9951 (SIP) set on the CME site can only make calls when the video is turned "OFF" on the set. When the video is turned "ON", the call does not establish and gets disconnected immediately after dialing the remote number.

    Note that a CME to CME internal video call is possible - the issue only happens when trying to make a call to the remote site.

    What config am I missing?!?!?!

    I am running 8.6

    TIA- Victor

    DB:3.18:H.323 Gateway Configuration Between Callmanager Express And Callmanager kf


    Hi Siva,

    SCCP IP PhoneCMEWANCUCMIP phone is OK

    SIP IP PhoneCMEWANCUCMIP phone is OK - with "Video" disabled on the set.

    SIP IP PhoneCMEWANCUCMIP phone - FAILS when "Video" is enabled on the set.

    I have configured Bandwidth and Location on CUCM

    Rgds

    - Victor

  • RELEVANCY SCORE 3.17

    DB:3.17:Ccm Trunk To A Router Running Cme m7



    can you create a trunk in CCM pointing to a router which is running CME? If so, then on the return side of that do you use just use normal destination pattern commands to point calls back towards the actual CCM?

  • RELEVANCY SCORE 3.10

    DB:3.10:Ccm Intergrate With Cme!!! dk



    CMM 4.13sr3c and CME3.3

    I am not using hardware gatekeeper at all.

    At CCM, I created Trunk(not gatekeeper controlled), and route group, list and pattern. I can make a call to CME's phone without any problem.

    At CME:

    voice class codec 1

    codec preference 1 g711ulaw

    codec preference 2 g729r8

    !

    !

    voice class h323 1

    h225 timeout tcp establish 3

    !

    dial-peer voice 1000 voip

    destination-pattern [12]...

    voice-class codec 1

    voice-class h323 1

    session target ipv4:10.0.0.21 (Sub)

    dtmf-relay h245-alphanumeric

    !

    dial-peer voice 1001 voip

    preference 1

    destination-pattern [12]...

    voice-class codec 1

    voice-class h323 1

    session target ipv4:10.0.0.20 (Pub)

    dtmf-relay h245-alphanumeric

    I could not call from CME to CCM, I got fast busy.

    Thanks for help.

    Ken

    DB:3.10:Ccm Intergrate With Cme!!! dk


    I forgot to put this command under interface:

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

    Thanks for help guys.

    KEn

  • RELEVANCY SCORE 3.06

    DB:3.06:Cme 2 Cme Voip Calls Work Every Other Time 9s



    have 2 cme's with checkpoint in between.

    cme A cme B

    ip phone on cme a call cue on cme b works every time

    ip phone on cme a calls ip phone on cme b it works every other time consistantly. One time ringback next time busy.

    phones in both locations are on different subnet from cue.

    i've tried to disable ip cef to no avail and still getting problem.

    debugs give disconnect cause code as network not available.

    any ideas as to why it would work one time and not the next?

    some sort of timer / timeout?

    thanks,

    paul.

    DB:3.06:Cme 2 Cme Voip Calls Work Every Other Time 9s


    have 2 cme's with checkpoint in between.

    cme A cme B

    ip phone on cme a call cue on cme b works every time

    ip phone on cme a calls ip phone on cme b it works every other time consistantly. One time ringback next time busy.

    phones in both locations are on different subnet from cue.

    i've tried to disable ip cef to no avail and still getting problem.

    debugs give disconnect cause code as network not available.

    any ideas as to why it would work one time and not the next?

    some sort of timer / timeout?

    thanks,

    paul.

  • RELEVANCY SCORE 3.06

    DB:3.06:Cme Voice Mail Profile sz



    Is there a way to emulate the CCM Voice mail Profile feature in CME? For example when extension 7005 calls 17001, it should be prompted with phone ext 7001 voice mail password. Can not use alternate extension in Unity as 7005 is a subscriber.

    Thanks

    John Arsenault

    Network Engineer

    Enventis Telecom

    DB:3.06:Cme Voice Mail Profile sz


    No, I don't think there is a way to emulate the CCM Voice mail Profile feature in CME

  • RELEVANCY SCORE 3.06

    DB:3.06:Send Message From Ccm To Cme 3z



    1) Is it possible to send message on Phone display connected to CME from CCM.

    Here is the scenario, there are 150 sites running CME and if I need to send a message to all 150 sites to be displayed on all the connected phones.

    How can it be done, if at all its feasible.

    Thanks

    DB:3.06:Send Message From Ccm To Cme 3z


    Request methods are XML structures that are passed to the AXL API server. The server receives the XML structures and executes the request. If the request completes successfully, then the appropriate AXL response is returned. REfer URL

    http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_programming_reference_guide09186a00801c5fab.html

  • RELEVANCY SCORE 3.05

    DB:3.05:Codec Between Voice Gateway And Ccm Runs Auto Attendent z8



    Hi All,

    I have CCM 4.1(3) running Auto Attendent Feature to answer Calls comming on FXO on Router 2800 working H.323

    I have to use codec G711 only between the VG and CCM to make the calls come on FXO hear the AA massege.

    How can I use codec G729 between VG and CCM and make AA Runs !!?

    Thanks

    DB:3.05:Codec Between Voice Gateway And Ccm Runs Auto Attendent z8


    This URL should help you:

    http://www.cisco.com/application/pdf/en/us/guest/netsol/ns268/c649/ccmigration_09186a008017bb4a.pdf

    http://www.cisco.com/warp/public/788/AVVID/g729a_codec.html

  • RELEVANCY SCORE 3.05

    DB:3.05:Ccm And Cme Comparison 3s



    There are some paper comparing features from Cisco CallManager to Cisco CallManager Express?

    tks

    DB:3.05:Ccm And Cme Comparison 3s


    Hi!

    While it's not a whole paper, you will find a table comparing CCME and CCM in "Selling IP Communications Express" at this links

    http://www.cisco.com/en/US/partner/products/sw/voicesw/ps4625/prod_brochure0900aecd8022483a.html

    kind regards

    bernhard

  • RELEVANCY SCORE 3.02

    DB:3.02:Video Call Between Cme And Cme Using H323 Trunk pp



    Dear Guy

    We having a problem using Video Advantage between CME and CME. We link CME between two site using H323 trunk.

    We able to make a call and all telephony  features between site but Camera is not pop-up on the screen. But if we use video call just only internal, there are no problem.

    Anyone know the configuration for using Video Advantage between Site using CME

    (both CME ISO is 12.4(24T) CME 7.1)

    Thank you

    DB:3.02:Video Call Between Cme And Cme Using H323 Trunk pp


    Dear Guy

    We having a problem using Video Advantage between CME and CME. We link CME between two site using H323 trunk.

    We able to make a call and all telephony  features between site but Camera is not pop-up on the screen. But if we use video call just only internal, there are no problem.

    Anyone know the configuration for using Video Advantage between Site using CME

    (both CME ISO is 12.4(24T) CME 7.1)

    Thank you

  • RELEVANCY SCORE 3.01

    DB:3.01:Ccm Cme Integration jd



    Hi,

    I connected CCM 4.1(3) and CME 3.3 and create an ICT in CCM.Also created 1 voip dial-peer in CME for calls to CCM

    dial-peer voice 7 voip

    destination-pattern 7....

    session target ipv4:192.168.1.75(CCM IP)

    codec g711ulaw

    Calls from CCM to CME is working fine. But when i am trying to call from CME phones to CCM phones it is getting busy tone.

    Also I have 1 CUE in CME Router. VoiceMail system is working properly from CME phones. But when i call VoiceMail number from CCM phones, it is getting busy tone.

    Pls help

    Thanks........


    DB:3.01:Ccm Cme Integration jd


    u welcome

    and thanks for Rating :)

  • RELEVANCY SCORE 3.00

    DB:3.00:Ccm - Cme - Cue Call Path Troubles. p7



    I've successfully intergrated CME and CUE and can make calls both ways between CME and my CCM cluster. However, when I call from CCM to CME and expect voicemail to pick up, my call gets dropped. I can call directly from CCM to CUE, so I know its not a route pattern issue. My dial-peers and telephony-service entries must be correct or I wouldn't be able to access voicemail from my CME phones. I've even tried configuring all possible "voice service voip" allow-connections entries between h323 and sip just in case. Any ideas? Thank you!

    CUE version = 2.2

    CME version = 3.4

    CCM version = 4.1

    router = 2621XM "c2600-adventerprisek9-mz.124-4.T1.bin"

  • RELEVANCY SCORE 3.00

    DB:3.00:Cme And Ccm Corporate Directory xz



    Hi,

    i have a setup where CCM can call CME and the other way around.

    the customer need is to have on the CCM a service pointing to the CME showing the users in the Corporate directory.

    also we need the other way around from CCM to see the CME CD.

    Good day

    DB:3.00:Cme And Ccm Corporate Directory xz


    Rob, I'm sorry to bother you, but I'm having the same issue, but unfortunately not the same result.

    I have a CCM 4.1(3) sr2 in the US and a CME in Japan.

    I'm trying to figure out how to read the directory from CME as an additional directory listing in CCM and then how to get the CME directory as an additional directory to CCM.

    I've tried editing the xmldirectory.asp on the CCM but it's not working, the path I'm using to connect to CME is http://10.103.0.2/localdirectory and I'm getting an XML Error [4]: Parse Error.

    Any ideas, would be greatly appreciated.

    you can e-mail me directly if you don't want to continue in this thread. dolah@vocollect.com

  • RELEVANCY SCORE 2.99

    DB:2.99:Cme And Gatekeeper 9m



    What is required in order to be able to call CME phones via a GK? I have CCM trunk, NetMeeting, and CME h.323 GW registered with GK with zone prefix zone2 4* where CME DNs are in the 4XXX range, I can call CCM phone from CCME phones via GK, but I cannot do that the other way around? What am I missing?

    Chris

    DB:2.99:Cme And Gatekeeper 9m


    Shanky and Michael,

    Your responses were really helpful, I got it to work, but I need to practice more with GKs and CME. I tried to rate your posts a while back and even today and it does not work. Not sure what the issue is.

    Chris

  • RELEVANCY SCORE 2.99

    DB:2.99:Conference Between Cucm 8 And Cme x9



    Hi voice team!

    I have a CUCM (site A) with conference bridge working properly to internal calls and with another site C using CME.

    But, the conference between CUCM (site A) and CME (site B) is not working...

    The Site B configuration is bellow:

     

    voice-card 0

    dsp services dspfarm

    interface Vlan1

    description VLAN-IT7

    ip address 172.22.0.1 255.255.255.0

    ip flow ingress

    ip nat inside

    ip virtual-reassembly in

    ip tcp adjust-mss 1452

    h323-gateway voip interface

    h323-gateway voip bind srcaddr 172.22.0.1

    sccp local Vlan1

    sccp ccm 172.22.0.1 identifier 10 version 7.0

    sccp

    !

    sccp ccm group 1

    associate ccm 10 priority 1

    associate profile 1 register confe1

    !

    dspfarm profile 1 conference

    codec g729br8

    codec g729abr8

    codec g729ar8

    codec g729r8

    codec g711alaw

    codec g711ulaw

    maximum sessions 2

    associate application SCCP

    telephony-service

    sdspfarm units 10

    sdspfarm tag 1 confe1

    conference hardware

    max-conferences 4 gain -6

    thanks.

    DB:2.99:Conference Between Cucm 8 And Cme x9


    Ah ok. I need to chenge the WAN codec for g711. I ´m going to test this..

    thanks

  • RELEVANCY SCORE 2.98

    DB:2.98:Cme One Way Video pm



    Hi!

    We have a follow equipment:

    Cisco 3825 with running CCME 4.1, CCM 4.2(3), MCU 3845, phones 7961.

    I'm trying to build meet-me video conference with ccm and ccme participants but there is only voice without video from the ccme part. Direct video calls from ccme to ccm are OK.

    Anybody knows what problem could be here?

    thanx

    DB:2.98:Cme One Way Video pm


    The reason may be VT advantage software when running on the PC needs to communicate with the phone. If the phone load on the phone does not include video support then the VT advantage software will not recognize the phone.

    Troubleshooting Cisco VT Advantage:

    http://www.cisco.com/en/US/docs/video/cuva/1_0/administration/guide/vtrble.html#wp1041576

  • RELEVANCY SCORE 2.97

    DB:2.97:Cme- Phone Remote Site zk



    Hi.

    I configured a CME in site B and configured trunk h323 with the Pabx Siemens in site A.

    My customer want two phones IP registered in CME of site B in site A.

    The phones are registered and do calls between them. But, with phones of PABX just the called listen, the calling (ip phones) don't listen.

    There are same configuration to cme support remote ip phones?

    Thanks

    DB:2.97:Cme- Phone Remote Site zk


    The first thing you should check when there is one way audio call, is routing.

  • RELEVANCY SCORE 2.96

    DB:2.96:Ccm - Cme Transfer Not Working k1



    Hi,

    i have scenario where CCM is configured with CME using ICT-non GK controlled.

    ip phone on cme calls ccm ip phone 1 and the ip phone 1 tries to transfer to ip phone 2 in ccm this does not work.

    I am wondering where i need to look to troublshoot this issue.

    CME:4.1 and CCM:4.3

    Good day

    DB:2.96:Ccm - Cme Transfer Not Working k1


    why u dont make the CME as H323 gateway on the CCM and make the proper dial-peers on CME

    ?

    According to Cisco press:

    When there is a call transfer for an H.323 call from one Cisco CallManager IP phone (phone A) to another (phone B), the H.323 signaling path does not change. It remains terminated on the Cisco CallManager server. The Cisco CallManager server establishes a new SCCP signaling path to phone B. Of course, the media path also has to change. Changing the media path on the IP phones is easy. The Cisco CallManager simply sends the appropriate SCCP messages to phone B, telling it to participate in the media connection to the external H.323 endpoint

    To change the media connection on the H.323 side, the Cisco CallManager uses a mechanism known as Empty Capabilities Set (ECS). This mechanism informs the external H.323 device that it should stop sending its media packets to phone A's IP address and should instead send the media packets to phone B's IP address. This mechanism allows the media stream to be redirected to the transfer-to destination phone while preserving the original H.323 control path connection

    MTP that can provide transcoding services. One example of this type of MTP is the digital signal processor (DSP) farms that are supported on Cisco IOS voice-enabled routers. DSP farms are controlled by a Cisco CallManager (or Cisco CME) using the SCCP protocol. The term transcode means the ability to convert the media stream from one codec type to another. You may sometimes see this term abbreviated as xcode.

    The drawback to this approach is the impact on overall scalability, because an MTP channel is needed for every H.323 (external) call

    good luck

    Please, if helpful Rate

  • RELEVANCY SCORE 2.96

    DB:2.96:Sip Trunk jk



    hi guys

    i have a gateway setup for testing as cme with sip trunk for pstn calls all good

    i want to test it with CCM

    if define it as normal h323 gateway will work normally or do i need to change any option

    CCM--h323---gateway--sip--ISP/pstn

    and if do it as bellow what is the binefit over the previous one !!!

    CCM --sip--gateway--sip--ISP

    thank you

    DB:2.96:Sip Trunk jk


    Nicholas is correct in his statements. CM 4.1 does a poor job in talking h323 to SIP. In most cases you will need an MTP to establish the calls. If you are low on MTP resources or none exist, the call may not complete or supplementary services will not work.

  • RELEVANCY SCORE 2.95

    DB:2.95:Gatekeeper Alias Static Command With Priority cz



    I am using Gatekeeper to control calls between a CallManager cluster and CME. When using the alias static command in gatekeeper, calls to CCM phones load balance between the Sub and the Pub. I cannot find a way to implement a priority as you can with the zone prefix command. Anyone know how to implement priorities on the alias static command?

    Thanks!!!

    Adam

    DB:2.95:Gatekeeper Alias Static Command With Priority cz


    Thanks for the response. I was thinking there was not a way, but needed to ask.

    So now let's approach this scenario from the other side. The Gk has an alias static from a remote H323 device (CME in the case). Since you cannot register CME to the Gk because he has an alias static, how do you get calls from CME into the GK destined for other sites?

    Thanks!!!

    Adam

  • RELEVANCY SCORE 2.95

    DB:2.95:Cme With Tandberg 6000 1k



    hi all,

    for our branch we have a cisco ISR router 2811running CME wih E1 port   and tandberg 6000 .

    my question is : what are  the requirements to make calls between tandberg 6000 and other tandberg  equipement in the HQ using E1 interface ?

    thanks for your reply .

    DB:2.95:Cme With Tandberg 6000 1k


    No additional licenses are necessary.

    Please remember to rate useful posts clciking on the stars below.

  • RELEVANCY SCORE 2.94

    DB:2.94:Call Forward/Call Transfer Between Cme And Ccm kf



    I have two phones with ext 2000 and 2001 respectively registered with CME 3.1. phone with ext 3000 registered with CCM 3.33. CME has dial-peer to CCM to reach 3000. Calls between them are good. Now I start to test call forward and transfer. According to CME 3.1 document we need to configure supplementary-service h450.12 and allow connection h323 to h323 the call can be transfered or forwarded from CME to CCM by hairpined since CCM is not H450.2/3 awared. But actually I didn't config these command at all. 2000 call 2001 and transfer or forward to 3000 works great. Any comment on this? Thanks. The router is 3745 with c3745-ipvoice-mz.123-8.T3.bin.

    DB:2.94:Call Forward/Call Transfer Between Cme And Ccm kf


    Thanks. I'll test this scenario again(ccm-cme-ccm). Now talking about the other forwarded scenario here CME1 to CME2 and forwarded to H323, this is same as on the first link. I also have no allow h323 to h323 configured. Call forward still work. If that's the case, it'll be different as the document, right. I'll do another test again later on. Thanks again.

  • RELEVANCY SCORE 2.93

    DB:2.93:Conferencing Problem 7f



    I have a CCM installation and a CME installation which are trunked. They use a g729r8 across the WAN.

    I establish a call between the sites no problem. They will use g729r8. From the phone registered on CCM, I then try to conference in a user in the same site (CCM) and when I try to conference, it doesn't work. It usually will drop the user in the CME site.

    Does anyone know exactly what I require setup on the CCM side to make this work? I have transcoding working fine on the CME router as I can 4 digit dial from CM to there then get into their voicemail (CUE) no problem. I'm running 4.0 CCM. Or at least where to start troubleshooting.

    DB:2.93:Conferencing Problem 7f


    I have a software conferencing bridge setup (one of the callmanagers). I also have a 2821 router setup as a transcoder. It registers with CCM OK and everything. Anything else I may be missing?

  • RELEVANCY SCORE 2.93

    DB:2.93:I Need To Configure Cluster Of Cme. Is It Possible? cx



    I need to configure cluster of CME. Is it possible?

    In CCM the modality does exist Publisher/Suscriber and in CME I do know SRST, however, is a model publisher/Suscriber, possible for CME?

    Regards,

  • RELEVANCY SCORE 2.93

    DB:2.93:Interclustertrunk Between Cme 3.4 And Ccm 3.3(4) 3f



    Hello,

    I´ve configured a Interclustertrunk between a Callmanager Express 3.4 and a Callmanager 3.3(4).

    A Call from a CME-Phone to a CCM-Phone, works without any problems.

    A call from a CCM-Phone to a CME-Phone results in a Call-Setup, that is disconnected after one or two ringback tones.

    The only hint I found was in Callmanager Trace for MTP:

    VoiceStrm|kDeviceDriverError - IP voice media streaming device driver error. Error:MTP: AId 1, CId 16777237, PId 17085265: Can't Mix Codecs.

    However I´m using G.711 ulaw at both sites.

    What could be the reason for this behavior?

    Thanx for any hint.

    Geri

    DB:2.93:Interclustertrunk Between Cme 3.4 And Ccm 3.3(4) 3f


    When you configured the ICT did you check "MTP required", make sure it is unchecked. Is the ICT configured correctly? Is the IP address correct? Is the Partition applied to the route pattern pointing to the route pattern listed in the CSS of the phone?

    Chris

  • RELEVANCY SCORE 2.92

    DB:2.92:Cme Sip To Ccm Calls Are Not Working zf



    Hi,

    I have implemented CME 7.0 and have registered both SCCP as well as SIP Phones, this gateway is registered to gatekeeper. Calls from SCCP CME to CUCM is working fine, but when i dial from the CME SIP phones to CCM Phone, it disconnects after ringing and does not allow to answer.

    I am able to dial from CCM to CME irrespective of protocol being used on the phone.

    Expert advise needed.

    Thanks,

    Vinay

    DB:2.92:Cme Sip To Ccm Calls Are Not Working zf


    Hi, check if you have this command enable in your CME.

    voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323

    hth

    David

  • RELEVANCY SCORE 2.91

    DB:2.91:Call-Forward Problem Between Cme And Panasonic Tda pa



    Hi,

    I have a trunk between a panasonic-tda and CME. all calls are routed succesfully and both can call each other but when an extension in CME side has forwarded his/her extension to another number, the calls from the panasonic to that extension in CME are failed. I will be appreciated if anybody could provide me any help in this regard.

    Thanks, Amir.

    DB:2.91:Call-Forward Problem Between Cme And Panasonic Tda pa


    Hi,

    unfortunately i'm not sure about the type of the trunk we are using. We've just setup a "voice dial-peer" in cme to route special calls (accroding to their leading number) to an IP address which is the IP-gateway card in the panasonic side. This card is installed and configured on a Panasonic PBX TDA-200 model and everything is working fine except call forwarding in cme side!

    Thank you very much.

  • RELEVANCY SCORE 2.90

    DB:2.90:Cme To Cme, Background Noise On Calls One Wan ps



    i have a CME in SiteA and another CME in SiteB, when call made across site, we experience noise and i have checked the codec used which is G.729. please what might be the problem.

    DB:2.90:Cme To Cme, Background Noise On Calls One Wan ps


    Hello

    i have checked the QOS SRND guide but didnt find what suit my need, please can i get an example config for the IP WAN QOS for point to point link below 768, do i need to enable cRTP on the link?

    please help as this noise is getting too much, thanks.

  • RELEVANCY SCORE 2.90

    DB:2.90:Ccme -- Ccm Voip Calls 3c



    Hi,

    I am trying to configure VoIP dialling between CallManager Express and CallManager. On a phone registered to the CCM i am able to VoIP dial to a phone registered to the CME, however i am NOT able to dial from CME phone to CCM phone. I have the following dial-peer configured:

    dial-peer voice 5001 voip

    preference 1

    destination-pattern 88390....

    session target ipv4:CCM

    dtmf-relay h245-alphanumeric

    codec g711ulaw

    no vad

    !

    I am trying to dial the number 883905700

    I have a translation pattern set up in CCM for 883905[5-9]XX

    Can anyone shed some light onto this issue for me?

    DB:2.90:Ccme -- Ccm Voip Calls 3c


    Try this command

    voice service voip

    allow connections h323 to h323

    h323

    post your results

  • RELEVANCY SCORE 2.90

    DB:2.90:Ccm 4.1 And Gatekeeper Cac cc



    I am using a gatekeeper to manage call admission control between CCM and CME. I have a Call Manager cluster in one zone and CME in another. I am limiting the interzone bandwidth for the CME zone, no limit on the CCM zone. Calls from CME phones seem to be requiring 16kbps per call (per the show gatekeeper zone status command) if I limit the bandwidth to 16 I can only place one call, 32 two calls etc. But from the Call Manager I can't place any calls to the CME zone unless I bump the bandwidth to 128. I can't seem to find any documentation on how much bandwidth a CCM (g.729) call requires vs a CME (g.729) call. Any help would be appreciated.

    DB:2.90:Ccm 4.1 And Gatekeeper Cac cc


    The calls are the other way around. The BRQ is relevant only if an ip phone on CME makes the call (or originates from CME). Then you need to set the BRQ parameter in the CM Service to TRUE.

    However, if you read the original post you'll notice the call is originating from CCM not CME.

    Paul

  • RELEVANCY SCORE 2.89

    DB:2.89:Sip Trunk Between Cucm 8.5 And Cme Calling Outbound Fails m9



    Hello all,

    I am trying to setup a SIP Trunk between CUCM 8.5 and CME.  Calling outbound to CUCM is failing with a code of SIP /2.0 503 Service Unvailable.  I can receive inbound calls from the CUCM.

    Here is the call flow.

    Phones(third-party)--SIP--2901(CME) --SIPtrunk--CUCM--MGCP--PSTN

    DB:2.89:Sip Trunk Between Cucm 8.5 And Cme Calling Outbound Fails m9


    Hi Matthew,

    The CUCM server is requesting that the gateway authenticate using Digest Authentication.

    You have a couple of options. If you don't want to use Digest Authentication on this trunk, you can uncheck the "Enable Digest Authentication" tick box on the SIP trunk security profile assigned to the SIP trunk used by CME. (Make sure that this profile is not used by any other SIP trunks first)

    if you need to use Digest Authentication, create an application user e.g "cmeTrunk" and assign a password in the "Digest Credentials" box then on your CME router add the following commands:

    sip-ua

    authentication username cmeTrunk password

    Dave

  • RELEVANCY SCORE 2.89

    DB:2.89:Transcoding Issue On Cme/Cue js



    Topology is as follows

    CCM 4.1(3) ----- H.323 Trunk ---- CME/CUE

    Setup to go g.729 of the trunk and then transcode to g.711 to go to CUE. Calls into CME work find but when forwarding to CUE we get a fast busy. If we configure it to go g.711 all the way the call works fine. The transcoder is configured on the CME but it does not seem to be working. Attached is the config. Any help would be appreciated.

    DB:2.89:Transcoding Issue On Cme/Cue js


    Hi all

    By default IOS does only route calls from voip dial-peers to pots dial-peers. To allow connections between 2 voip dial-peers you need to put above commands into the config.

    In your scenario this must be enough:

    voice service voip

    allow-connections h323 to sip

    Hope this helps.

  • RELEVANCY SCORE 2.87

    DB:2.87:How Does Callmanager Express (Cme) Work? xc



    Hello,

    Does anyone know how CME works with loca calls? My understanding was that CME uses Skinny to talk to CME and RTP was the call is set up. When I perform the commands show sccp connections and show voip rtp connections detail it shows that no sccp and rtp connections are active. Any clue?

    On the other end when I run CME remotely between 2 separate call managers across a WAN. I get a RTP connection, an H323 connection, but no SCCP...any clue on this.

    thanks for the help.

    DB:2.87:How Does Callmanager Express (Cme) Work? xc


    I haven't had my morning coffee, so my ramblings may not make sense. But I'll have at it.

    For internal (on-net) calls, once CME call control has identified and connected the calling and called endpoints, RTP and state information is exchanged between IP telephone endpoints. Voice traffic does not traverse CCME, as CCME just sets up the call and gets out of the way.

    For external (off-net) calls, any IOS supported voice endpoint can be used to connect a call off a gateway. It may go H.323 or SIP to another gateway, to head to another site or the PSTN. The CME router would then maintain the call control as it is converted.

    Hope this helps.

    Pat

  • RELEVANCY SCORE 2.87

    DB:2.87:Cme And Ccm 7p



    I need to call between a CCM cluster to a CME site. I can get it working using gatekeeper for call routing.

    Any idea how to get this done without gatekeeper. On CME router, I will create dial-peer pointing to CCM, correct? From CCM, how do I send a call to CME?

    DB:2.87:Cme And Ccm 7p


    Right. However, the only thing that would not work correctly is ringback. So, if your users can live without ringback then you should be OK...

    -Jaret

  • RELEVANCY SCORE 2.87

    DB:2.87:Cube Supplementary Services Problem !!! d7



    hi to all..

    we have the following setup..

    CUCM8.0 cluster ( using ICT-nonGK ) CUBE ( 2911 with 15.1(3T) IOS ) CUCM 8.5 cluster ( using ICT-nonGK ) and vice versa..

    it's a classic CUBE configuration for h.323 to h.323 using ICTs and in future h.323 to sip and vice versa call setup..

    after these two clusters we're going to introduce few more CUCM clusters and few 3rd party SIP PBXes..

    so, basic call setup works just fine..

    but calls requiring hold or transfer do not..

    after the calls is established and somebody tries to transfer call or put someone on hold, the call drops or we loose signaling and cannot resume the call..

    when enabling MTP on ICT in CUCM, everything works just fine..that's in case of CUCM software MTP..

    even when we register CUBE MTP to the CUCM and use it also as MTP on ICT it works fine..

    but, it's not the propper solution..

    we don't want to be forced to introduce MTPs in the scenario, where CUBE should do all the work..at liest we wouldn't like to..

    any thoughts?..why CUBE cannot handle supplementary services between two CUCM 8.x clusters..

    here is the partial CUBE config with relevant info for call setup from CUCM8.0 to CUCM 8.5..

    transocode is registered locally to CUBE and MTP is registered to CUCM 8.5..it just in testing purposes..

    *****************

    voice service voip no ip address trusted authenticate mode border-element allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none h323  emptycapability sip  early-offer forced  midcall-signaling passthru!  

    voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 g729r8 codec preference 4 g729br8!

    !voice class h323 1  h225 timeout tcp establish 3 

    !

    !interface GigabitEthernet0/0 ip address X.X.X.X duplex auto speed auto h323-gateway voip interface h323-gateway voip bind srcaddr X.X.X.X

    !

    !sccp local GigabitEthernet0/0sccp ccm CUCM8.5 SUB identifier 3 version 7.0sccp ccm CUCM8.5 SUB identifier 2 version 7.0sccp ccm CUBECME identifier 1 version 7.0sccp ip precedence 3sccp!sccp ccm group 2 associate ccm 2 priority 1 associate ccm 3 priority 2 associate profile 3 register cube_mtp_cucm!sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register cube_xcode_cme switchback method graceful!dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 10 associate application SCCP!dspfarm profile 2 mtp codec pass-through maximum sessions software 1000 associate application SCCP! 

    !dial-peer voice 38 voip description ****** Incoming calls  ****** modem passthrough nse codec g711alaw redundancy incoming called-number XXX voice-class codec 1 dtmf-relay h245-alphanumeric rtp-nte fax rate disable no vad!

    !dial-peer voice 3810 voip description ****** Outgoinging calls SUB ****** translation-profile AAA destination-pattern XXX session target ipv4:CUCM8.5SUB voice-class codec 1 voice-class h323 1 dtmf-relay h245-alphanumeric no vad!

    !dial-peer voice 3811 voip description ****** Outgoinging calls to PUB ****** translation-profile AAA preference 2 destination-pattern XXX session target ipv4:CUCM8.5PUB voice-class codec 1 voice-class h323 1 dtmf-relay h245-alphanumeric no vad

    !telephony-service sdspfarm units 5 sdspfarm transcode sessions 10 sdspfarm tag 1 cube_xcode max-ephones 1 max-dn 1 ip source-address CUBE-CME-IP port 2000 max-conferences 8 gain -6 transfer-system full-consult!

    *******************

    thanx..

    regards..

    DB:2.87:Cube Supplementary Services Problem !!! d7


    Hi Sinisa

    Glad to hear that the issue is resolved.

    ICT uses some non standard H245 messages. This may be for Supplementary services and call control. Like for ICT trunk, you do not have option to control the H245 negotiation. Because TCS negotiation doesn't work same way as any other H245. The advantage of this, it will save you some MTP resources. Also, on ICT trunk, you can control the MoH type (user hold, network hold), also audio type. But dis advantage is, the nonstandard messages.

    Unless you have that command configured, CUBE would drop it. And that is why your supplementary service wasn't working.

    Thats all I can remember at this point.

    Thank you

    - Abu

  • RELEVANCY SCORE 2.87

    DB:2.87:Trunk Between Ccm And Ccme pm



    I have a CCM in Site A and I have a CME in Site B. I have a VPN link connecting them.

    I need 4 digit dialing between them.

    How do I integrate them. I tried creating an intercluster trunk but it did not work.

    DB:2.87:Trunk Between Ccm And Ccme pm


    you probably need the following command (PIX):

    fixup protocol h323 ras 1718-1719

    although for VPN I dont think it is required, since the traffic is not NAT'd over VPN

  • RELEVANCY SCORE 2.86

    DB:2.86:Cant Make Phone Call To Cme Over Saf. j8



    Hi Guys,

    We have a plan to deploy SAF in our company, so I did some test in the past a few weeks.

    The result list below: (CME 8.1, CCM 8.0.3)

    1. CME -- SAF FW -- CCM.                        Call from CME to CCM is successful.

    2. CCM -- SAF FW -- CME.                        Can't call from CCM to CME. CME router can receive H323 or SIP signaling, but call reject.

    3. CME -- SAF FW -- SAF FW -- CME.     Can't call between CME. CME router can receive H323 or SIP signaling, but call reject.

    I thougt it is the CME router SAF inbond dial peer problem. I can't set up the SAF inbond dial peer cause it is the dynamic incoming dial peer.

    From Cisco document, "Voice Service Advertisement Framework Feature":

    xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx

    The patterns you configure are prepended with a site code before being advertised, creating an incomingdynamic dial peer with the translation rule to remove the site code from the dialed number.

    For example:site-code : 333voice translation-rule 2000000001rule 1 /^333\(...\)/ /\1/

    A block of dial-peer tags in the range of 1073741824–2147483647 are used by Voice SAF. The staticdial-peer tag range is lowered to 1–1073741823. The dynamic incoming dial peers described in theprevious example are created using a tag in the Voice SAF range.Only one incoming dial peer is created for a trunk route advertising SIP and H.323 signaling. Similarly,translation rule tags in the range of 1073741824–2147483647 are used for Voice SAF.

    xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx

    Someone can help me?

    Many thanks,

    Jeffrey

    DB:2.86:Cant Make Phone Call To Cme Over Saf. j8


    Guys,

    I had similar experience until I allowed CUCM ip under my trusted authenticated list

    voice service voip

    ip address trusted list ipv4: x.x.x.x y.y.y.y

    x.x.x.x.=cumc ip address..

    If you do a debug voip ccapi and ccsip messages you should get a disconnect cause of 21. That will confirm that its the ip address trust list issue

    Please rate useful posts

    "I am complete in God, God completes me"

  • RELEVANCY SCORE 2.86

    DB:2.86:Calliung Ccm 4.2 And Cme Locations kk



    I trying to determine the best method to route calls between a CCM 4.2 and CUE location. I have a Intercluster Trunk to route calls to the CME site via an MPLS network. In the event of a failure I want to end the calls over the PSTN. What would be the easiest method, AAR, multiple, Route Patterns, other? Your input is appreciated.

    DB:2.86:Calliung Ccm 4.2 And Cme Locations kk


    If the H323 is already in another RG or selected under a route pattern, it cannot be used again. You need to take it out from any other RG or RP to use it.

    HTH

    java

    if this helps, please rate

    www.cisco.com/go/pdihelpdesk

  • RELEVANCY SCORE 2.86

    DB:2.86:Call Pickup Group Notification On Cme 7k



    Dear all,

    On CME, can I use the call pickup notification feature like on CCM?

    I want to use the visual alert when we have calls in the pickup group.

    thanks,

    DB:2.86:Call Pickup Group Notification On Cme 7k


    I'm afraid the CME doesn't have that feature. You can use hunt-groups or shared lines to make sure you are always have a visual/audible alert.

  • RELEVANCY SCORE 2.86

    DB:2.86:Cme To Cucm 9z



    For the love of me, I cannot get my CME to dial out to a CUCM, or even full version CCM. I know it's rather simple. I created a dial-peer that looks something like this:

    dial-peer voice 25 voip

    destination pattern 1...  ---This accounts for any phones that start with 1 and have 4 digits total

    session target ipv4:192.168.1.252   ----IP Address of my CCM or CUCM

    codec g711ulaw

    no vad

    I have created a route pattern on both the CCM or CUCM and can get calls from either to CME, but not the other way around. As soon as I finish the 4th number of an extension for either system (i have switched between both systems thinking it was some config error on either one), I get the fast busy signal. I simply don't get it. Anyone have any ideas? Thanks in advance for any help.

    Rob

    DB:2.86:Cme To Cucm 9z


    I will remember that. Thanks again!

  • RELEVANCY SCORE 2.86

    DB:2.86:Cme 4.02 - No Conference Call 97



    I have a 3825 with CME4.02 that will not allow me to establish conference calls. After I connect to one person and press Confrn, the phone displays "No Line Available". I have max-conferences 12 gain -6 set.

    Thanks,

    Jay

    DB:2.86:Cme 4.02 - No Conference Call 97


    Sometimes, the simplest things.... Dual-line fixed it. Luckily, I have a DB report routine to redo the config!

    Thanks, Brandon!

    Jay

  • RELEVANCY SCORE 2.86

    DB:2.86:Call Unhold Fails Between Cme And Ccm Over Ict cp



    We have a remote CME site (v.3.3) and a central CCM site (4.1.3sr4d). The sites use an ICT for calls between the two sites. Calls between the two sites are successful. However, if there is a call between the central site and the remote site and the central site (CCM) puts the remote site call on hold, then tries to resume the call, both phones (central and remote) act as if the call is still on hold. After approximately 10 seconds the call is disconnected. I've had an open case with TAC for three weeks now (two different engineers) and they can't give me a solution. According to them the debugs and traces that we have run indicate that the call is clearing normally.

    Any help that anyone can provide is greatly appreciated.

    DB:2.86:Call Unhold Fails Between Cme And Ccm Over Ict cp


    Thanks for the kind words and great to be of help.

    EDIT: On a side note, even I'm with TAC. We do the best we can, except sometimes, we are humans :-)

  • RELEVANCY SCORE 2.85

    DB:2.85:Not Able To Make Calls From Cme To Ccm d9



    Hi All,

    I have two sites one is main site with CCM and other is a remote site with CME, the CME is connected to the router at mainsite, i am able to ping back and forth, I have ICT configured in the CCM to make calls to CME, this is working properly, ( I can call from CCM to CME ), but I am not able call from CME to CCM, I get a bus signal, I have a VOIP dial peer configured in the CME pointing directly to CCM.

    CCM - 172.200.1.10

    CME - 192.4.1.1 ( loop back interface )

    serial between CME ( 10.20.20.2 ) at remote site to Router ( 10.20.20.1 ) at main site, RIP is the routing protocol used. This is carried out in LAB.

    CME:

    allow-connections h323 to h323

    dial-peer voice 6000 VOIP

    destination-pattern 6....

    session target ipv4:172.200.1.10

    no shut

    I have configured allow-connections a the main site Router as well.

    Below is the debug output when I made call from CME ( 5XXX extension series ) to CCM ( 6XXX extension series ) I called from 5001 to 6002

    *Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:   Match Rule=DP_MATCH_ANSWER; Calling Number=5001*Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:   Is Incoming=TRUE, Number Expansion=FALSE*Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:   Dial String=, Expanded String=, Calling Number=5001T   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH*Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:   Result=-1*Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:   Match Rule=DP_MATCH_ORIGINATE; Calling Number=5001*Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:   Is Incoming=TRUE, Number Expansion=FALSE*Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:   Dial String=, Expanded String=, Calling Number=5001T   Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH*Apr  8 04:49:48.140: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:   Result=Success(0); Incoming Dial-peer=20001 Is Matched*Apr  8 04:49:48.144: //-1/774322318048/DPM/dpMatchCore:   Dial String=6, Expanded String=6, Calling Number=   Timeout=FALSE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH*Apr  8 04:49:48.144: //-1/774322318048/DPM/MatchNextPeer:   Result=MORE_DIGITS_NEEDED(1); Outgoing Dial-peer=1*Apr  8 04:49:48.144: //-1/774322318048/DPM/dpMatchCore:   Result=1*Apr  8 04:49:48.344: //-1/774322318048/DPM/dpMatchCore:   Dial String=60, Expanded String=60, Calling Number=   Timeout=FALSE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH*Apr  8 04:49:48.344: //-1/774322318048/DPM/MatchNextPeer:   Result=MORE_DIGITS_NEEDED(1); Outgoing Dial-peer=1*Apr  8 04:49:48.344: //-1/774322318048/DPM/dpMatchCore:   Result=1*Apr  8 04:49:48.544: //-1/774322318048/DPM/dpMatchCore:   Dial String=600, Expanded String=600, Calling Number=   Timeout=FALSE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH*Apr  8 04:49:48.544: //-1/774322318048/DPM/MatchNextPeer:   Result=MORE_DIGITS_NEEDED(1); Outgoing Dial-peer=1*Apr  8 04:49:48.544: //-1/774322318048/DPM/dpMatchCore:   Result=1*Apr  8 04:49:48.748: //-1/774322318048/DPM/dpMatchCore:   Dial String=6002, Expanded String=6002, Calling Number=   Timeout=FALSE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH*Apr  8 04:49:48.748: //-1/774322318048/DPM/MatchNextPeer:   Result=Success(0); Outgoing Dial-peer=1 Is Matched*Apr  8 04:49:48.748: //-1/774322318048/DPM/MatchNextPeer:   Result=Success(0); Outgoing Dial-peer=6000 Is Matched*Apr  8 04:49:48.748: //-1/774322318048/DPM/dpMatchCore:   Dial String=6002, Expanded String=6002, Calling Number=   Timeout=TRUE, Is Incoming=FALSE, Peer Info Type=DIALPEER_INFO_SPEECH*Apr  8 04:49:48.748: //-1/774322318048/DPM/MatchNextPeer:   Result=Success(0); Outgoing Dial-peer=1 Is Matched*Apr  8 04:49:48.748: //-1/774322318048/DPM/MatchNextPeer:   Result=Success(0); Outgoing Dial-peer=6000 Is Matched

    Please help to resolve

    thanks

    Shaggy

  • RELEVANCY SCORE 2.85

    DB:2.85:Qos For Voip s8



    We have 3MB link between 3745 router runing CME 3.1 and CCM 3.3 behind 3640 router IOS 123-11.T.

    Round trip time from CME to CMM is 250ms.

    Looking for documents or/and links to configure best QoS over this 3 MB link.

    DB:2.85:Qos For Voip s8


    Here is something to get you started.

    http://www.cisco.com/en/US/netsol/ns340/ns394/ns171/ns109/networking_solutions_white_paper09186a008018913f.shtml

    http://www.cisco.com/en/US/tech/tk543/tk759/technologies_tech_note09186a0080094bc3.shtml

    http://www.cisco.com/en/US/tech/tk543/tk889/tech_protocol_family_home.html

    Hope this helps.

    Please remember to rate all replies

  • RELEVANCY SCORE 2.85

    DB:2.85:Sip Trunk Between Ccm 5.0 And Softswitch xj



    I need to establish a sip trunk between CCM 5.0 and Softswitch for IP phone to be able to call PSTN via softswitch. How to verify whether the SIP trunk between both devices is UP and running?

    DB:2.85:Sip Trunk Between Ccm 5.0 And Softswitch xj


    I need to establish a sip trunk between CCM 5.0 and Softswitch for IP phone to be able to call PSTN via softswitch. How to verify whether the SIP trunk between both devices is UP and running?

  • RELEVANCY SCORE 2.85

    DB:2.85:Cac On Cme a1



    Hi People

    I have a CME with n CUVA on 2 sites. I want to limit the bandwidth to permit 2 or 3 video calls between sites. I have endpoint tandberg that I don't want limit video calls between their!

    It's possible to create samething like CAC on CME?

    Thanks

    DB:2.85:Cac On Cme a1

    Hi,

    Cisco IOS software has a built-in CAC mechanism with the call threshold interface command. This feature limits both inbound calls and outbound calls for a specific interface on the Unified CME router once a maximum threshold has been exceeded. For example, the following command causes calls from GigabitEthernet0/0 to be rejected after the number of simultaneous inbound/outbound calls exceeds five. Calls are then allowed once the maximum number of simultaneous calls falls below 3.

    call threshold interface GigabitEthernet0/0 int-calls low 3 high 5

    The benefit of this feature is that it does not require gatekeeping and can operate across multiple dial-peers.

    Best Regards

  • RELEVANCY SCORE 2.85

    DB:2.85:Cme With Wan kj



    Hi,

    i am in the process of Pre-sales, where a small customer has an HQ with CME and he has 8 intl branches.

    All the branches are connected through VPN for Data transfer, solution provided is to install an IP phone in each location and register the IP Phone with the CME in the HQ. it works perfect, my concern is the QOS, i am wondering if there is a way to limit the number of calls within the CME or consider the remote branches as regions like in the CCM.

    Good day

    DB:2.85:Cme With Wan kj


    Not yet, but I think will be in CME 4.3. The new phones (models ending with 5) have it, the older ones do not.

    Thanks for the nice rating and good luck!

  • RELEVANCY SCORE 2.84

    DB:2.84:Ccm W/ Cme Remotes (In Lieu Of Srs) 87



    Can CME router can be substituted for an SRST router, whereby phones register with the centralized CCM cluster and secondarily with CME. Driving this design is the ability for our services to be initiated from a CME with direct egress to the ISP node (w/o the call path traversing thru an MTP.

    DB:2.84:Ccm W/ Cme Remotes (In Lieu Of Srs) 87


    The problem is that customer wants everything to be centralized, so the simple solution would be to deploy Call Manager in HQ, and run SRST in remote locations, but We will be providing IP Trunking (SIP Trunking) to all remote locations that can only in remote locations in a CME router, but then customer loses the benefits of Centralized Management.

    I thought the best option would be is to have phones registered via Centralized Call Manager, but still use CME Router to route call to SIP Cloud.

    I hope am making sense.

    Regards,

  • RELEVANCY SCORE 2.84

    DB:2.84:Cme-Ccm aa



    I have an IP link between a CME and a CCM and I can call the CME from a phone in the CCM but not the way around.

    I tried with both g711 and g729 unsuccesfully.

    DB:2.84:Cme-Ccm aa


    first you must have one of the connection options from ccm to cme: ict or h225 trunk. on the cme you must have voip dial-peers in direction of ccm. at least you have to check, the css and significant digit field of the gateway. very important, the ip address of the ict trunk or h225 trunk in ccm must be the h323 source interface from cme.

    greetings

    mehmet

  • RELEVANCY SCORE 2.84

    DB:2.84:Routing Calls From Cme To Ccm Through A Gatekeeper Testing With Csim Start zk



    I am trying to route calls from cme to ccm through a gatekeeper. Everything seems to be working fine except when I test the calls from the router using csim start I cannot get the IP phone on the ccm side to ring. I have tech prefix set to 1# on ccm h323 trunk and 2# on cme. I have tech prefix set on and session target ras on dial-peer pointing to the gatekeeper. I have a translation pattern set in ccm to strip the 1# off of the called number from cme. when I do a show h323 gateway I see the no response from user count increment everytime I make a csim start test call. When I do a debug gatekeeper main 10 I can see where the tech prefix is matched and the zone is matched. Nothing happens after that. By the way calls from ccm to cme work. Just not the other way around.

    DB:2.84:Routing Calls From Cme To Ccm Through A Gatekeeper Testing With Csim Start zk


    Do the calls work if a CME phone is calling?

    A few general things to check:

    1) You do not have a separate "Gateway" configuration in CCM for the CME. The trunk is sufficient.

    2) If you do have a Gateway configuration, and you cannot remove it, make sure that its CSS includes the partition where the translation pattern is located.

    3) Make sure the trunk is using the correct CSS as well.

    4) Make sure you have no codec mismatch issues causing the h245 portion of setup to fail. It's possible that the inbound dial-peer in CME is not the same as the outbound, so codec selection by CME could be different depending on the call direction.

    5) Along those lines, are there any resulting CAC/bandwidth constraints?

    On CME, debug h225 asn1, debug ras, and debug h245 asn1 are your verbose logging friends. Use them in off-peak times.

    On CCM, enable "Arbitrary" debugging levels for the callmanager service on the subscriber(s) who could be receiving the calls. Then the Q.931 translator to verify that the call is received and get the disconnect cause, and use notepad to find out why.

    Let me know if this helps get you pointed in the right direction by rating the post.

    Michael

  • RELEVANCY SCORE 2.84

    DB:2.84:Iphone Cisco Mobile 8 Integration With Cme ss



      I have set up Cisco 8 Mobile for iPhone and am able to call between the iPhones DNs and Desk phone DNs but how do I  enable ability to see on an iPhone when I have an external call coming in? Also I am unable to make make external calls, does the iPhone use the same dial-peers?

    Thanks for your help. : )

    DB:2.84:Iphone Cisco Mobile 8 Integration With Cme ss


    Yup I meant mobility, I just read that Jabber is the latest instance of Cisco Mobility. I will test that and see how I get on. How do I get my external calls thou to the iPhone using jabber?

  • RELEVANCY SCORE 2.84

    DB:2.84:Cme And Gatekeeper 39



    Hi all,

    I have 2 cme system in two locations .I want to use one cme router as gatekeeper .And limit calls between the sites to 4 g729 calls .Can any one show how can i do this defining zones?

    thanks in advance

    Thusharam

    DB:2.84:Cme And Gatekeeper 39


    This is meant for a lab just to get the concept down. Down the road may use it for larger systems.

  • RELEVANCY SCORE 2.83

    DB:2.83:Cme, Call Forward To Cue From Ccm Ip Phone pk



    I want to call forward the call from CCM IP phone to CME ephone's voicemail which setup in CUE. works okay between CME ephones. configured voice service as follows but no luck. what did I missing to implement?

    voice service voip

    allow-connections h323 to h323

    allow-connections h323 to sip

    allow-connections sip to h323

    allow-connections sip to sip

    no supplementary-service h450.2

    no supplementary-service h450.3

    -CCM4.1.3, configured H225 trunk. leave uncheck the MTP on the trunk device

    -gatekeeper to connect between CCM and CME

    -CME3.3, h323 to gatekeeper and sip to CUE

    -CUE2.1

    Thanks in advance,

    DB:2.83:Cme, Call Forward To Cue From Ccm Ip Phone pk


    It works by restart the CME router and have a question the sip-ua output. I have two media streams but the 2nd shows "STREAM_IDLE". I think this is for g729 connected to CCM via h323 gk. Can I get an explanation why?

    CME#sh sip-ua calls

    SIP UAC CALL INFO

    Call 1

    SIP Call ID : 1B6D4D7C-FB4E11DA-802BAF13-A036A662@10.253.66.254

    State of the call : STATE_ACTIVE (7)

    Substate of the call : SUBSTATE_NONE (0)

    Calling Number : 4083132006

    Called Number : 4211

    Bit Flags : 0x101A0030 0x100000 0x500

    CC Call ID : 95

    Source IP Address (Sig ): 10.253.66.254

    Destn SIP Req Addr:Port : 10.253.66.2:5060

    Destn SIP Resp Addr:Port: 10.253.66.2:5060

    Destination Name :

    Number of Media Streams : 2

    Number of Active Streams: 1

    RTP Fork Object : 0x0

    Media Stream 1

    State of the stream : STREAM_ACTIVE

    Stream Call ID : 95

    Stream Type : voice-only (0)

    Negotiated Codec : g711ulaw (160 bytes)

    Codec Payload Type : 0

    Negotiated Dtmf-relay : inband-voice

    Dtmf-relay Payload Type : 0

    Media Source IP Addr:Port: 10.253.66.254:16998

    Media Dest IP Addr:Port : 10.253.66.2:16904

    Orig Media Dest IP Addr:Port : 0.0.0.0:0

    Media Stream 2

    State of the stream : STREAM_IDLE

    Stream Call ID : -1

    Stream Type : voice+dtmf (1)

    Negotiated Codec : No Codec (0 bytes)

    Codec Payload Type : 255 (None)

    Negotiated Dtmf-relay : inband-voice

    Dtmf-relay Payload Type : 0

    Media Source IP Addr:Port: 10.253.66.254:17120

    Media Dest IP Addr:Port : 0.0.0.0:0

    Orig Media Dest IP Addr:Port : 0.0.0.0:0

    Number of SIP User Agent Client(UAC) calls: 1

    SIP UAS CALL INFO

    Number of SIP User Agent Server(UAS) calls: 0

    CME#sh sccp connections

    sess_id conn_id stype mode codec ripaddr rport sport

    1 2 xcode sendrecv g711u 10.253.66.254 2000 16518

    1 1 xcode sendrecv g729 10.253.66.254 2000 17620

    Total number of active session(s) 1, and connection(s) 2

  • RELEVANCY SCORE 2.83

    DB:2.83:Sip Trunk Between Ccm 4.X And Uc500 zs



    Hi,

    It's possible to create a sip trunk between a CCM 4.x and a UC500 in order to use the CCM as pstn gateway and permit internal voip calls?

    Thanks

    Mirko

    DB:2.83:Sip Trunk Between Ccm 4.X And Uc500 zs


    Hammad1980 wrote:

    What about If I'm usnig MGCP?

    thanks,

    Only use H.323 or SIP for trunking.

  • RELEVANCY SCORE 2.83

    DB:2.83:Configure Srst Fallback Support On Cme 19



    Hi,

    Don't quit understand this,

    telephony-service

    srst mode auto-provision none

    srst dn line-mode dual

    max-ephone 10

    max-dn 10

    ip source-address IP of CME

    end

    Does it mean:

    I do not need to configure ephone and ephone-dn, as the config will be download from the CCM in time of working condition.

    When the connection broke, it will fallback to the config that it download from CCM previously.

    What is the difference between this and call-manager-fallback ?

    DB:2.83:Configure Srst Fallback Support On Cme 19


    If you require only straight forward fallback, where phones simply require the ability to receive calls and make external calls, then CME SRST is not the option.

    In this particular instance configure SRST 'call-manager-fallback'.

    With auto-provision all, the ephone-dn's and ephones are created in the running configuration when they re-register. IP Phones will not fallback if you have not configured the SRST reference with CallManager and enabled SRST on the voice-gateway.

    If you enable only CME/SRST without registering the gateway within CCM and without specifying a SRST reference for the device pool then you will have to configure the gateway for H323 and full CME, ensuring that you provision TFTP, ringtones, firmware, and both ephone(mac-addresses) and ephone-dn's.

    As I have mentioned, SRST Auto-provision all ensure that the mac-addresses are created, and the ephone-dn associate with the mac is inserted into the running configuration.

    Prior to this the phones are dependant on CCM for obtaining their firmware and configuration. In the event of SRST, the phones have already loaded their configuration and firmware from CCM, and therefore simply re-register with their existing configuration.

    Regards

    Allan.

  • RELEVANCY SCORE 2.83

    DB:2.83:No Ringback Tone When Ccm Ip Phones Call Cme Ip Phone kp



    THe CCM 3.3 has set up the H323 Gateway and it is pointing at the IP address of the CME. When IP Phone of CCM system calls IP Phone at CME, the IP Phone at CCM does not have ringback tones but the call can be established.

    On the CME router, where can I put those progress_ind alert command as I cannot put any commands on those auto-generated dial-peer voice 2000X pots for the ephones.

    Thanks,

    ---

    Raymond

    DB:2.83:No Ringback Tone When Ccm Ip Phones Call Cme Ip Phone kp

    Hi,

    the guys below are not correct. The outside dial-tone and incoming called-number have nothing to do with the ringback.

    Login to your Cisco router and run "debug cch323 h225". Make a call and see if any ALERTING messages comes to the router.

    CallManager has a problem where it does send ALERTING message, but with no PI (Progress Indicator) inside.

    You have three choices:

    1. Put the "tone ringback alert-no-pi" in the pots dialpeer.

    2. Put progress_ind setup 1 and progress_ind progress 3 at the pots dial-peer.

    3. Check in the CCM if the ringback is set to H225 UserInfo.

  • RELEVANCY SCORE 2.82

    DB:2.82:Cme - Dailing Out As Quick As Possible 9s



    We use CME. When users attempt to place an outbound call, they only have approximetly four seconds to complete the 10 digit dailing. Beyound four seconds, a busy signal is generated, thus, unable to place the call.We obviously need to make an adjustment, thus, giving more time to the Users, to successfully place calls. Any insight on CME configuaration would be helpful and very much appreciated. Thanks!

  • RELEVANCY SCORE 2.82

    DB:2.82:Cme 3.3 Via H323 To Ccm4.1.3 cs



    I'm trying to route calls via H323 between CME and CCM.

    I can't figure out why the call fails from the phone on the CME site to the CCM. I get a fast-busy.

    Obviously I'm missing something.

    From the CCM I can call the phone on the CME.

    Any advice will be appreciated please. I've attached the router config.

    DB:2.82:Cme 3.3 Via H323 To Ccm4.1.3 cs


    TAC resolved the problem; over complicated configuration.

    Kill the voice service hairpin routing. This seems to be the key.

    Bind h323 in the interface. I'd taken it out.

    I never tried binding the interface w/o the hairpinning.

    Configure standard dial-peer(s) and telephony service.

    Most of the over complication came from trying out the QTC.

    Thanks TAC.

  • RELEVANCY SCORE 2.82

    DB:2.82:Uccx For Cme 98



    Hello,

    We have a CME 4.1 and we need to add some of the Contact Center services to our enviroment:

    Microsoft CRM.Call queue and load balancing calls between agents.

    I need to clear few points after reading the UCCX SRND:

    Do we need the CRS with every UCCX installation?Do we need the IVR for Call queue?Can we install the UCCX, IVR and CRS all in one server?Does the UCCX for CME support the above requirement or we shall go for the CCM?

    Regards,

    Zak

    DB:2.82:Uccx For Cme 98


    Thank you both for your feedback,

    I have been using the UCCX 7.9 BOM tool, and i found Voice XML ports for DTMF, ASR ports and TTS ports; since we are using the voice gateway to terminate the PSTN calls on and we have only one location do we need the Voice XML ports for DTMF, ASR ports and TTS ports?

    Regards,

    Zak

  • RELEVANCY SCORE 2.81

    DB:2.81:Distinctive Ringing k3



    how to distinctive ringing to differentiate between internal and external calls in CCM 3.3

    thank you

    DB:2.81:Distinctive Ringing k3


    For the same ring tone you can set different cadence for internal and external calls.

    You can have Distinctive ring (on-net vs. off-net)

  • RELEVANCY SCORE 2.81

    DB:2.81:How To Troubleshoot Dropped Calls m1



    hi, i am experiencing drop calls between remote sites

    Backgound: the network between a HQ and 2 remote sites are connected in a hub-and-spoke. the HQ is using CCM5.1, while the remote sites are using CME. Remote sites are connected by trunks to the HQ ccm.

    Problem: calls between remote sites are dropped. when the called party answer the call using softkey, the call didnt established. however, the phone received another ringing tone almost immediately. when the called party answer the phone again, the call is dropped permanently with the calling party hearing the hangup tone.

    there are no problems with calls between HQ and either remote site.

    appreciate any advice.

    cash

    DB:2.81:How To Troubleshoot Dropped Calls m1


    Hi Cedric,,

    Thanks for explaining, that was really helpful, I was just going through Mr.cashqoo attachment in that i was bit confused, could please help me out with it.

    I couldn't understand why was the output showing Called Number=3076 which was suppose to be calling number and at the end it shows Called Number=7387552 which is proper.

    Oct 19 09:12:38.615: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:   Calling Number=, Called Number=3076, Peer Info Type=DIALPEER_INFO_SPEECHOct 19 09:12:38.615: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:   Match Rule=DP_MATCH_DEST; Called Number=3076Oct 19 09:12:38.615: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:   Result=Success(0) after DP_MATCH_DESTOct 19 09:12:38.615: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:   Result=SUCCESS(0)   List of Matched Outgoing Dial-peer(s):     1: Dial-peer Tag=20004     2: Dial-peer Tag=20072Oct 19 09:12:38.615: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:   Calling Number=, Called Number=7387552, Peer Info Type=DIALPEER_INFO_SPEECHOct 19 09:12:38.619: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:   Match Rule=DP_MATCH_DEST; Called Number=7387552

     

  • RELEVANCY SCORE 2.81

    DB:2.81:Sip Trunk Between Cme 7.0 And Ccm 4.1 9j



    hi, i am trying to make a sip trunk between a cme 7.0 and ccm 4.1, the ccm 4.1 already has a sip trunk working to an OCS, so i created a new one, but i had to change the ccm's incoming port to 5061, to create it.....!!so i want to know, how can i configure the cme to point the sip trunk (dial-peer sip), to the ccm's 5061 port.

    DB:2.81:Sip Trunk Between Cme 7.0 And Ccm 4.1 9j


    This command will also change the port used to send SIP traffic.

    Hope this helps.

    Brandon

  • RELEVANCY SCORE 2.81

    DB:2.81:Cme: Ctl-Signing And Security p8



    Hello

    With CME we have to sign the CTL-File for security with two SAST-records (similiar to the 2 security tokens with CCM). These records are generated by the CME-router.

    Now, what if i have to replace the CME-Router, how can i export the 2 SAST-records to the new router, so that i don't have to delete the CTL-file manually on each phone?

    Other Question:

    With 2 secure CMEs how are the encryption keys being exchanged between the 2 routers?

    DB:2.81:Cme: Ctl-Signing And Security p8


    Hello

    With CME we have to sign the CTL-File for security with two SAST-records (similiar to the 2 security tokens with CCM). These records are generated by the CME-router.

    Now, what if i have to replace the CME-Router, how can i export the 2 SAST-records to the new router, so that i don't have to delete the CTL-file manually on each phone?

    Other Question:

    With 2 secure CMEs how are the encryption keys being exchanged between the 2 routers?

  • RELEVANCY SCORE 2.80

    DB:2.80:Srst Question jz



    We have a CCM (hq and branch site) with one gateway setup in each location as h323.

    If CCM goes down and phones register to gateways for SRST, I need some help understanding the call flow when CCM is down.

    Example, in normal condition our voip dial-peer points the 4 digit destination-pattern to CCM. But when CCM goes down and those phones register to gatekeeper, how do incoming calls know to "not" go to CCM? Is it because dial-peer will fail to CCM and phones that registered now have "virtual" pots dial-peers for each one that will allow calls to be routed appropriatly to those phones?

    What about if one uses CME for SRST. Phones/dial-peers are already created. So how would calls correctly get routed to CCM in normal condition and only to CME registered phones only when CCM is down? Is that becuase even though the ephone ephone dn's were created (which creates virtual pots dial-peers), since the phones are "not connected" the call will go to ccm via voip dial-peer that points to it?

    Thanks

    DB:2.80:Srst Question jz


    Worth to point this out:

    After phone fail to CCM and then register to VGW, VGW will query phone configuration and also build the database of all newly registered phone configuration.

    I belive, this means all phones ext. numbers (say 4 digits) will be recognize by VGW (through 'virtual' dail peers) and in turn vgw will route the incoming calls properly.

    If you use TranslationPattern in CCM to manipulate the incoming DID number, then you need to have similar way in VGW using 'num-exp', 'tranlation rule', etc,.

    Hope the above will explain the questions.

    Please let us know your test result.

  • RELEVANCY SCORE 2.80

    DB:2.80:Alerting Name Presented Between Cme And Cucm 37



    Hi All,

    I have a customer who is currently on a CME system, they then purchased a CUCM.

    The CUCM is running in parallel to the CME, with a small number of users migrated from CME to CUCM.

    The CME is configured as a h323 gateway in the CUCM.

    There are dial-peers in CME to point to the CUCM for those users that have been migrated.

    Calls between the systems are fine.

    Is there a way of sending or receiving the alerting names of the users on the different systems? An extra command under the voip dial-peers perhaps?

    When I call between 2 CMEs I see the alerting names between them.

    Regards

    Felix

    DB:2.80:Alerting Name Presented Between Cme And Cucm 37


    Is "Display IE Delivery" checked on the H323 GW in CUCM?

    Chris

  • RELEVANCY SCORE 2.80

    DB:2.80:Software To Try Cti Ports ? f8



    Hi,

    I have some problems integrating a third party unified messaging solution with a Cisco CallManager 4.1.3.

    This UM Solution is using CTI ports to connect to CCM. CTI Ports are well registered but I'm unable to establish incoming and outgoing calls.

    I verified everythings (partition, CSS, user association, ...) but without success.

    So I'm searching to test CTI Ports without the 3rd Party UM Solution.

    Something like a CTI Phone that would be able to register CTI Ports on CCM and to make/receive calls.

    Does it exist ? If yes, were can I find it ?

    Thanx,

    Bastien.

    DB:2.80:Software To Try Cti Ports ? f8


    hello,

    I'm trying to use the CTI ports to receive and make calls, but with no success. (using Jtapi to monitor those calls).

    So, to test those cti ports, I'm using a cisco softphone, but I cannot able to detect any lines (I configured already a cti port in CallManager and saw the troubleshooting in the cisco's documents).

    Could you help me?

    I'm working with CallManager 4.2

    This is my email: pedro.albuquerque@mbcpbank.com

    Thank you!

  • RELEVANCY SCORE 2.79

    DB:2.79:Cme To Callmanager Via Gatekeeper W/O Tech Prefix Or Default Tech fp



    Anyone know if it's even possible for CME to communicate with CallManager over an gatekeeper controlled H.225 trunk WITHOUT a Tech Prefix of any kind configured on either CM or CME?

    I can't seem to get it working when trying to call a Callmanager IP Phone from CME (3XXX extension range on CM).

    Note: If I add a technolgy prefix of say 1# to both CM and CME (including "tech-prefix 1#" on dial-peer), everything works fine. But I don't want to use a tech prefix!! Anyone know if this is possible? Or does CallManager require a tech-prefix in order for it to receive inbound calls??

    PS. I know the basics are working fine, since calls work when a tech prefix is assigned to CM and to the dial-peer on CME.

    When I do a DEBUG H225 ASN on the gatekeeper, I'm told the call is rejected because the called number is not registerd with the gatekeeper. Since CallManager never registers it's numbers with a Gatekeeper is a prefix the only way this thing works??

    BTW. Calls from CM to CME work fine since CME registers it's E.164 no's with the gatekeeper.

    Here's my config....

    CallManager (CCM)

    ----------

    -No prefix configured...

    Gatekeeper CFG

    --------

    gatekeeper

    zone local CCIELAB1 compucom.com 21.1.1.1

    zone remote BACKBONE compucom.com 10.1.1.1 1719

    zone prefix CCIELAB1 3... gw-priority 0 CME

    zone prefix CCIELAB1 3... gw-priority 10 CCM

    zone prefix CCIELAB1 3* gw-priority 0 CME

    zone prefix CCIELAB1 3* gw-priority 10 CCM

    no shut

    CME Config

    ----------

    interface Loopback0

    ip address 121.1.1.1 255.255.255.0

    h323-gateway voip interface

    h323-gateway voip id CCIELAB1 ipaddr 21.1.1.1 1719

    h323-gateway voip h323-id newyork

    h323-gateway voip bind srcaddr 121.1.1.1

    dial-peer voice 3000 voip

    destination-pattern 83T

    session target ras

    dtmf-relay h245-alphanumeric

    gateway

    DB:2.79:Cme To Callmanager Via Gatekeeper W/O Tech Prefix Or Default Tech fp


    So here's the deal. H.323 gateways will register themselves w/ gatekeeper using their E.164 addresses. So if you have two CME's registered to a gatekeeper they'll be able to call each other w/o using any tech prefixes.

    CallManager, however does not register any of its DN's w/ the gatekeeper. So, the only way to route a call TO CallManager from the gatekeeper is using some form of a prefix. If you only need to place calls from a CCM cluster into a gatekeeper clound...no prefix is needed...only when calls are coming in the reverse direction.

    Thanks!

    Patrick

  • RELEVANCY SCORE 2.79

    DB:2.79:How To Remove 0 On Missed Calls - Ccm 7.1.5 xf


    Hi,
    we have ccm 7.1.5 and i added new sip trunk (new provider).
    When I have missed calls, numbers have one extra "0" in front of number so i am unable to directly call missed number.
    How can i remove that extra "0" that that don’t affect other SIP trunks configured on CCM.
     
    Regards,
    Ivan

    DB:2.79:How To Remove 0 On Missed Calls - Ccm 7.1.5 xf


    There is no CUBE. Today I contact the provider and they told me that thay send me without that extra 0.

     

  • RELEVANCY SCORE 2.79

    DB:2.79:Cube Sip Error s8



    Hi,

    I have a CUBE connecting one CME and one Avaya IPOffice via SIP trunks. Calls from CME can establish the call, but fail when the user at IPOffice picks up the phone. 'debug ccsip message' shows:

    the CUBE is sending 'BYE' because of

    CSeq: 102 BYE

    Reason: Q.850;cause=127

    Content-Length: 0

    Calls from the Avaya to CME are working fine.

    Any guidance to the problem will be appreciated.

    Thanks.

    DB:2.79:Cube Sip Error s8


    Thanks Nick,

    'Debug ccsip all' shows the codec in one leg is different from the other. I changed to the same codec and is working now.

    The command is really helpful.

    Thanks again.

  • RELEVANCY SCORE 2.79

    DB:2.79:Cme 3.2 Over Vpn Unable To Register zj



    Hi, I have setup a CME 3.2 system at work with 2 phones on can call between no problem.

    On the single switch at work both phones register successfully to the CME router and I?m able to place calls to and from each other. At another location I have an 1801 router with a point to point VPN to the office Pix.

    On the 1801 router I have set option 150 to point the CME router. When I plug the phone into the 1801, I get the following message on the CME router and the phone fails to register. The phone keeps on trying to register and the same message appears.

    Oct 4 09:14:35.457: %IPPHONE-6-REG_ALARM: Name=SEP000E386DCFEB Load=CP79050101S

    CCP030530B.zup Last=Initialized

    Oct 4 09:14:35.457: %IPPHONE-6-REGISTER: ephone-2:SEP000E386DCFEB IP:192.168.1.

    53 Socket:2 DeviceType:Phone has registered.

    Oct 4 09:14:36.475: %IPPHONE-6-UNREGISTER_ABNORMAL: ephone-2:SEP000E386DCFEB IP

    :192.168.1.53 Socket:2 DeviceType:Phone has unregistered abnormally.

    MY setup:

    192.168.103.X ------- 2950 Switch ----- CME Router

    |

    Pix External IP (DHCP for 192.168.103.X LAN)

    |

    IP Sec VPN

    |

    1801 External IP (DHCP for 192.168.1.X LAN with option 150 pointing at CME)

    |

    192.168.1.X

    Any pointers will be greatly appreciated.

    Thanks

    Craig.

    DB:2.79:Cme 3.2 Over Vpn Unable To Register zj


    I think tftp may be timing out, any ideas ?

  • RELEVANCY SCORE 2.79

    DB:2.79:Unity Express - Cue, Application Mode Callmanager Not Express xp



    Anyone know how to change the "application mode" of CUE to CME and not CCM? Is this based on the license? I have a CME and not CCM and somehow the config got set for CCM, even though I have done a clean install.

    DB:2.79:Unity Express - Cue, Application Mode Callmanager Not Express xp


    Thanks for the help guys, saved me a call to TAC.

    I ran through initialization and couldn't specify MWI or anything.  Got on-site, looked into it, CCM mode it is!

    Swapped, reloaded, onto more installation tasks.

    Cheers!!!

    JB

  • RELEVANCY SCORE 2.78

    DB:2.78:Extension Mobility Between Cme And Cucm xa



    Hello,

    We use CUCM on our main sites. We've started to deploy CME on field offices.

    Is it possible to activate extension mobility between CUCM and CME ?

    ie: being able to log on a CME managed phone with a CUCM account and from CME to CUCM ?

    Thanks,                 

    DB:2.78:Extension Mobility Between Cme And Cucm xa


    Hello,

    Thanks for your answer, I was hopping to have someone with a by-pass solution...

    Regards,

    NH

  • RELEVANCY SCORE 2.77

    DB:2.77:Trunk Between Callmanger 3.3.X And Cme 3.4 dp



    I´ve a Callmanager 3.3 on the one side and a Callmanager Express on the other side.

    I´ve configured a Intercluster Trunk between both systems as it is described in the Callmanager Express System Administration Guide.

    When I start a call from a phone connected to the CME to a phone connected to the CCM everything works always as it should.

    When I start a call from a phone connected to a Callmanager to the CME site I have 3 different situations: 1. sometimes the call setup is successfull and every thing works fine. 2. sometimes the callsetup is successfull but I´ve one way voice from CCM phone to CME phone

    3. sometimes the call setup is not successfull. It rings twice on the CME-Phone and then the call setup fails.

    Thanx for any idea,

    Geri

    DB:2.77:Trunk Between Callmanger 3.3.X And Cme 3.4 dp


    Hallo,

    I have similar problem, call setup is ok once, then call setup fails.

    Sometimes if I reset trunk in CCM I am successfull again , but most times connection from CCM to CME doesnt work, the other direction is always ok, do you find a solution for the problem ?

    br

    Michael

  • RELEVANCY SCORE 2.77

    DB:2.77:Gatekeeper/Gateway Co-Resident Problem zp



    This scenario took me quite a while and still no luck.

    One co-resident CME-A gateway/ H.323 gatekeeper in same cisco rtr, and one other CME-B gateway.

    ephones under CME-A cannot connect to CME-B via gatekeeper, CME-B ephone cannot talk to A vise versa,

    Each time, after dial digits, got busy signal.

    Troubleshooting:

    Both CME-A and CME-B registered to Gatekeeper. All of zone prefixs are also up.

    If without Gatekeeper, using IPIP gateway config, both CME-A and CME-B ephones can communicate.

    Routing between CME-A ephone and CME-B ephone been verified.

    The issue only happen when calls been routed through gatekeeper.

    I am afraid it is due to co-resident CME-GW/GK issue.

    IOS: c2600-ipvoice_ivs-mz.124-9.T7.bin

    Any thought?

    The config can be provided if needed.

    DB:2.77:Gatekeeper/Gateway Co-Resident Problem zp


    thanks all for input Finanlly, I kick off co-resident gw/gk and use dedicate gk.

  • RELEVANCY SCORE 2.77

    DB:2.77:Ccm Unable To Call Into Cme (Voip) j3



    Hi

    Connectivity

    CCM1---(h323)---HQRTR/IPIPGW----(H323)---CME(Branch2)

    Scenario 1:

    CME phones can call into CCM1 phones.

    Scenario 2: (Problem)

    CCM1 IP phones not able to dial into CME IP Phones. When CCM1 phones call into CME phones, CME phones will ring, however, upon picking up the call at CME phones, CCM1 phones will continue to ring. It's as if CME phones is either

    a) not telling CCM1 that he has picked up the call

    or

    b) some signalling that i've not configured.

    Please help.

    Thanks

    DB:2.77:Ccm Unable To Call Into Cme (Voip) j3


    Hi,

    which IOS are you running on the CCME ? I would try to upgrade it latest maintenace, but just a precaution, because at this point it seems the problem is on the CCM unable to recognize answer and proceed with the call.

    I know that is possible to take traces and decode on the CCM, unfortunately I'm unable to personally help in that, hope someone here can.

  • RELEVANCY SCORE 2.77

    DB:2.77:Ccm To H323 Gw To Gk To Cme Issue ms



    Hi,

    I`m a bit lost on a issue that i am experiencing right now.

    I have a Callmanager, whos is connected to a h323 GW, that GW is connected to a Gatekeeper, the Gatekeeper has another 2 CME locations connected.

    When i dial from a CME site to a CME site (through GK) the destination ip phone rings, and upon answer i have audio.

    When i dial from a CME site to the CCM Site (through the GK), the destination phone will ring, and upon answer i have audio.

    However, when i call from an IP Phone at the CCM site to a CME site (using the h323 GW and GK), the CME phone will ring, however when i answer the CME Phone, the cme phone produces several beeps (sounds like hold beeps), and the CCM Phone stays on ring out, and drops the call after 10 seconds

    now i`ve added the h225 connect passthru, and when i dial now the same happens, however the CCM IP Phone shows connected and still drops the call after 10 seconds.

    When i configure a trunk on the CCM to the Gatekeeper, everything works like it should, so it looks like my h323 GW hop is causing some issues.

    my guess is that some h225 h245 and probably h450 messages are missing between the h323 GW and the CME`s.

    Does anyone have idea?

    Grtz

    DB:2.77:Ccm To H323 Gw To Gk To Cme Issue ms


    Hello,

    Sorry for the delayed reply, the customer from the other end was using an E1 EM signaling with the PBX. We ended up changing the E1 card to ISDN E1 and the calls went through. As you can see it is an EM signaling issue but this was the fastest fix to the problem.

    Hope this helps,

    Regards,

  • RELEVANCY SCORE 2.77

    DB:2.77:Voip Cme To Cme Via Gre Tunnel 1j



    We have a HQ office and Branch office. Each have thier CME. Calls between the offices are VoIP across the Internet using a GRE tunnel.

    When employees from the HQ office visit the branch office and use IP Communicator, they are able to successfully register to thier CME across the GRE tunnel. However, when a Remote office worker visits HQ they are unable to successfully register thier IP Communicators to the Branch office CME across the GRE tunnel.

    Any ideas why this is so?

    Here is the debug from the Branch office CME when an IP Communicator located at the Main office network is trying to register:

    See Attachment

    DB:2.77:Voip Cme To Cme Via Gre Tunnel 1j


    Hi, I haven't looked at the debug, but generally these issues are caused by the phone not getting the "right" tftp file from the "right" CME, or general connectivity problems, eg, not every subnet is reachable from any other.

  • RELEVANCY SCORE 2.77

    DB:2.77:Ringback Between Cme And Call Manager 1k



    Hi

    I have a Call Manager 4 installed at HQ site with an H323 gateway handling the incoming calls via PRI ,i then have a CME at Branch site with the branch phones registering to the CME ,the calls for the branch phones come in via the PRI at HQ site and the Receptionist at HQ site then transfers the calls to the CME users ,the calls then go via VOIP Dial-peer from H323 GW to the CME. Problem is that there is no ringing tone heard by the HQ receptionist while the CME phone is ringing ,only silence ,and only once they answer she can hear them. Any ideas ?

    DB:2.77:Ringback Between Cme And Call Manager 1k


    Hi

    I have a Call Manager 4 installed at HQ site with an H323 gateway handling the incoming calls via PRI ,i then have a CME at Branch site with the branch phones registering to the CME ,the calls for the branch phones come in via the PRI at HQ site and the Receptionist at HQ site then transfers the calls to the CME users ,the calls then go via VOIP Dial-peer from H323 GW to the CME. Problem is that there is no ringing tone heard by the HQ receptionist while the CME phone is ringing ,only silence ,and only once they answer she can hear them. Any ideas ?

  • RELEVANCY SCORE 2.77

    DB:2.77:Integrating Cme With Ccm ka



    hi, i am integrating a cme v 7.0 with a ccm 4.1, using h323 protocol, in the cme, i created a dial-peer voip pointing to the ccm's ip add, and in the ccm, y create a gateway h323, pointing to the cme ip add, and then the route pattern with the specific gateway.

    the problem is, when an cme ip phone call to an ccm extension, it rings, the call establishes but after 4 sec the call is disconnected.

    if the call is from the ccm phone to the cme phone, the call works fine,

    the cause is temporary failure (41),

    do you know why does this cause happen?

    DB:2.77:Integrating Cme With Ccm ka


    Hi,

    do you have "codec g711u" under the DP for CCM? Which setting do you have on CCM for the CME GW ?

  • RELEVANCY SCORE 2.77

    DB:2.77:No Missed Calls Are Shown In 8.0(3) For 7912 a9



    Just found another bug for the too long list. The above defect showed after an upgrade. Downgraded to 8.0(2) and missed calls are OK.

    This is with CME, not sure if it happens with CCM too.

    DB:2.77:No Missed Calls Are Shown In 8.0(3) For 7912 a9


    Hi Paolo, did you have seen a similar issue with the ip phones 7941/61 in CUCM?, My problem is that the missed calls only shows the Voicemail port number. I'm using CUCM7+CUC7.

    Thanks a lot

    David

  • RELEVANCY SCORE 2.76

    DB:2.76:H323 Trunk Between Cucm And Cme c9



    Hi All,

    I'm having some issues setting up an H323 trunk between Call Manager 5.1 and CME 7.1. The CME router is connected back via a VPN tunnel within the router back to the ASA at HQ. So far I have setup the h323 gateway on the CCM using the IP of the CME router with a Route Pattern using that Gateway. On the CME side I setup the dial-peer and voice service voip as follows:

    dial-peer voice 5000 voip

    destination-pattern [5678]...

    session target ipv4:192.168.11.21

    dtmf-relay h245-alphanumeric

    codec g711ulaw

    no vad

    !

    !

    !

    voice service voip

    allow-connections h323 to h323

    allow-connections h323 to sip

    allow-connections sip to h323

    !

    When I call from CCM to CME, the calls go through fine, though from CME to CCM, I get a busy after about 10 seconds. When I do a debug voip ccapi inout, it shows that the call goes through the correct dial-peer, though I get an error cc_api_call_disconnected: Cause Value=38.

    I've been racking my head trying to figure this one out without any success, so any help would be greatly appreciated.

    Thanks,

    Ryan

    DB:2.76:H323 Trunk Between Cucm And Cme c9


    Check the IE option in the H.323 gateway configured in the CCM. Also in the CME verify the following:

    At the voice service level:

    voice service voip

    h323

    h225 timeout ntf 50-5000

    h225 display-ie ccm-compatible

    At the voice class level:

    voice class h323 1

    h225 display-ie ccm-compatible [system]

    h225 timeout ntf 50-5000

    interface Serial0/3/0:23

    no ip address

    encapsulation hdlc

    isdn switch-type primary-ni

    isdn incoming-voice voice

    isdn map address *. plan isdn type unknown

    isdn supp-service name calling

    no cdp enable

    You may need to upgarde the IOS in order to get access to those commands.

    MK

  • RELEVANCY SCORE 2.76

    DB:2.76:Ccm Transcoder Registration Question zk



    WE have been troubleshooting a complex issue by where transcoder is not being invoked for WAN calls. TAC has recommended we attempt to take the transcoder registration off callmanager, and put this transcoder solely registered to the h323 gateway. Is that possible? I know if we were running CME it would be done this way, but is it possible when using CCM with h323 gateway?

    Our problem is we need an MTP for supplementary services (other side doesn't support h323 v2). So we have the MTP checkbox selected on gateway page in CCM. Region between phones and software MTP is G711. Region between phones and MTP-Xcode is G711. Region between gateway and MTP's is G711. Gateway is set to use G729. Trace files reveal that the transcoder is not being invoked.

    Regards,

    DB:2.76:Ccm Transcoder Registration Question zk


    Hi Daniel,

    Could you please explain what the problem is? Since as per the case notes it seems that we just need an XCODE for the calls from the GW among the other regions; other than that, the GW must have a MRGL on CCM with a transcoder and software MTPs for it to support the call.

    Thanks and please keep us posted!

  • RELEVANCY SCORE 2.76

    DB:2.76:Cme To Cucm Hold Issue fz



    I have a small CUCM 6 setup that has one CUCM server. My CUCM server routes calls to a CME 4.1(0) router via h323 gateway configuration.

    I can make calls between locations without any issues.

    However when calls from the CME phones are placed on hold on by the CUCM phones, the CME phones get a busy signal and the call drops.

    This is also affecting call transfers of the CME calls between the CUCM phones.

    On the CME phones I can hold and transfer the CUCM calls without any issues.

    Has anyone any ideas how to solve this issue?

    DB:2.76:Cme To Cucm Hold Issue fz


    If your IP phones need an MTP and they're using G.729 the software MTP in CUCM won't do this. That's a possible cause, but from speculation I can't say what the problem may be with that.

    -nick

  • RELEVANCY SCORE 2.76

    DB:2.76:Video Calls Between Cme 7.1 And Cucm 7.1 ax



    Hello,

    I need advice.

    There are two geo sites connected via Internet. First site contains CME(cisco2901)  another site contains  ASA + ISR 2811+CUCM. On first site, cisco ip phones registered on CME on second site on the CUCM.

    I need to setup video calls(VT Advantage) between sites. I plan to perform it with sip trunk between CME and CUCM. Will it work or maybe i need consider another solution?

    Thanks in advance

    DB:2.76:Video Calls Between Cme 7.1 And Cucm 7.1 ax


    I know this is a very old post, but I want to thank Vishal for the above.  I was troubleshooting one way video problem between CME and CUCM.  The asymmectric payload full command fixed it.

  • RELEVANCY SCORE 2.75

    DB:2.75:Ccm-Ipipgw-Cme zd



    Hi,

    i have the setup above

    CCM calls CME: ccm ITC Non-GK to ipipgw using g711 and ipipgw to cme sip g729

    CME calls CCM: cme to ipipgw h323 g729 and ipipgw to ccm sip g711 sip trunk

    why question is where to do the stripping because of in-band/out-band dtmf and how can i test if this is working or not.

    dtmf-relay rtp-nte digit-drop h245-alphanumeric

    Good day

    DB:2.75:Ccm-Ipipgw-Cme zd


    Hi,

    i have the setup above

    CCM calls CME: ccm ITC Non-GK to ipipgw using g711 and ipipgw to cme sip g729

    CME calls CCM: cme to ipipgw h323 g729 and ipipgw to ccm sip g711 sip trunk

    why question is where to do the stripping because of in-band/out-band dtmf and how can i test if this is working or not.

    dtmf-relay rtp-nte digit-drop h245-alphanumeric

    Good day

  • RELEVANCY SCORE 2.75

    DB:2.75:Voice Traffic Between Cme And Asterisk zd


    A custumer is asking if it would be possible to make calls between CME and asterisk via VPN.

    Any of you have tried this scenario?

  • RELEVANCY SCORE 2.75

    DB:2.75:Unable To Establish Calls Between Cme And Ccm. 8z



    Hi All,

    I have a lab setup having CME and CCM. IP Phones are registered to both the Call agents and have registered CME as H323 gateway in CCM. I am able to call from IP phone registered with CCM to CME but not vice versa. Any idea why so? IP Phone on CME's number is 64058XXX and IP Phone on CCM is 6XXX

    I have attached running config of CME below.

    GW-2#sh run

    Building configuration...

    Current configuration : 2710 bytes

    !

    version 12.4

    service timestamps debug datetime msec

    service timestamps log datetime msec

    no service password-encryption

    !

    hostname GW-2

    !

    boot-start-marker

    boot-end-marker

    !

    !card type command needed for slot 1

    no logging console

    enable password cisco

    !

    no aaa new-model

    !

    resource policy

    !

    no network-clock-participate slot 1

    !

    !

    ip cef

    no ip dhcp use vrf connected

    ip dhcp excluded-address 10.10.50.1 10.10.50.10

    !

    ip dhcp pool CME

    network 10.10.50.0 255.255.255.0

    default-router 10.10.50.1

    option 150 ip 10.10.50.1

    !

    !

    !

    voice-card 0

    no dspfarm

    !

    voice-card 1

    no dspfarm

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    !

    interface GigabitEthernet0/0

    no ip address

    duplex auto

    speed auto

    !

    interface GigabitEthernet0/0.4

    encapsulation dot1Q 4

    ip address 10.10.20.1 255.255.255.0

    no snmp trap link-status

    !

    interface GigabitEthernet0/0.5

    encapsulation dot1Q 5

    ip address 10.10.40.1 255.255.255.0

    no snmp trap link-status

    !

    interface GigabitEthernet0/0.6

    encapsulation dot1Q 6

    ip address 10.10.50.1 255.255.255.0

    no snmp trap link-status

    !

    interface GigabitEthernet0/1

    no ip address

    shutdown

    duplex auto

    speed auto

    !

    interface Serial0/0/0

    ip address 192.168.1.2 255.255.255.252

    no fair-queue

    clock rate 125000

    !

    interface Serial0/0/1

    no ip address

    shutdown

    clock rate 125000

    !

    interface Serial0/1/0

    ip address 192.168.1.5 255.255.255.252

    !

    ip route 10.10.10.0 255.255.255.0 192.168.1.1

    ip route 10.10.10.0 255.255.255.0 192.168.1.6 200

    ip route 10.10.30.0 255.255.255.0 192.168.1.1

    ip route 10.10.30.0 255.255.255.0 192.168.1.6 200

    !

    !

    ip http server

    no ip http secure-server

    !

    !

    !

    tftp-server flash:/PHONE/7940-7960/P00308000500.bin alias P00308000500.bin

    tftp-server flash:/PHONE/7940-7960/P00308000500.loads alias P00308000500.loads

    tftp-server flash:/PHONE/7940-7960/P00308000500.sb2 alias P00308000500.sb2

    tftp-server flash:/PHONE/7940-7960/P00308000500.sbn alias P00308000500.sbn

    !

    control-plane

    !

    !

    !

    !

    !

    !

    !

    dial-peer voice 1 voip

    destination-pattern 6...

    session target ipv4:10.10.20.100

    !

    !

    !

    telephony-service

    load 7960-7940 P00308000500

    max-ephones 52

    max-dn 192

    ip source-address 10.10.50.1 port 2000

    create cnf-files version-stamp Jan 01 2002 00:00:00

    max-conferences 8 gain -6

    !

    !

    ephone-dn 1 dual-line

    number 8000 secondary 64058000

    !

    !

    ephone-dn 2 dual-line

    number 8001 secondary 64058001

    !

    !

    ephone-dn 3 dual-line

    number 8002 secondary 64058002

    !

    !

    ephone-dn 4 dual-line

    number 8003 secondary 64058003

    !

    !

    ephone 1

    mac-address 000D.654B.8D36

    type 7960

    button 1:1 2:2

    !

    !

    !

    ephone 2

    mac-address 001D.728E.D42C

    type CIPC

    button 1:3 2:4

    !

    !

    !

    line con 0

    line aux 0

    line vty 0 4

    password cisco

    login

    !

    scheduler allocate 20000 1000

    !

    end

    Regards

    Mangesh A. Bhende

    DB:2.75:Unable To Establish Calls Between Cme And Ccm. 8z


    Hi,

    I have added h323-gateway voip bind srcaddr ip-address command, under interface configuration mode and the problem got resolved.

    Regards

    Mangesh A. Bhende

  • RELEVANCY SCORE 2.75

    DB:2.75:Ccm To Cme Voicemail Issue jz



    Hi all,

    I have set up CCM 4.1(3) to talk to CME via a ICT (GK) trunk. Everything works ok except calls from a CCM phone over the ICT trunk do not get successfully forwarded to CUE (the call just drops). I can dial the VM pilot number direct from the CCM phones, it just appears to be when the CME phones forward on no answer or busy.

    When i call from a PSTN phone the CME phone transfers to CUE fine.

    I have checked codecs and everything is G711u for now to avoid any transcoding issues.

    Any help would be greatly appreciated.

    Many thanks,

    Gary.

    DB:2.75:Ccm To Cme Voicemail Issue jz


    Almost a year later and your solution helped me. +5 for you. Thanks for the assistance.