"CCM SIP Trunk" task issue

Hello,



I am trying to complete the "CCM SIP Trunk" task in Voice Lab Workbook 1 and I have a strange issue when I place a call from PSTN to BR1 : the phones rings OK but when I "answer" at BR1, the call ends and re-rings immediately. When I "answer" again, it ends definitively... See below my "debug ccsip events" output.

VORack07R2#
Apr 14 08:31:35.309: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
Apr 14 08:31:35.313: //56/7FDFC1A8801A/SIP/Event/sipSPICreateRpid: Received Octet3A=0x80 -> Setting ;screen=no ;privacy=off
Apr 14 08:31:45.157: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Apr 14 08:31:45.161: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP

Apr 14 08:31:45.161: //57/7FDFC1A8801A/SIP/Event/sipSPICreateRpid: Received Octet3A=0x80 -> Setting ;screen=no ;privacy=off
Apr 14 08:31:53.205: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
VORack07R2#



1/ I noticed the following (see below) in SRND 4.X concerning SIP trunks and MTP, which could explain why it is not possible to answer the call after ringing.

SRND 4.X and SIP Trunks : (http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42trunks_ps556_TSD_Products_Implementation_Design_Guide_Chapter.html)

Support for SIP trunks was first added in Cisco Unified Callmanager Release 4.0.
...
In Cisco Unified CallManager 4.2 and earlier releases, SIP trunks have the following characteristics:
•...
•Each call using a SIP trunk requires a media termination point (MTP) resource.


2/ There was already a topic concerning this issue at http://ieoc.com/forums/t/4986.aspx but there isn't a definitive explanation...


3/ If there's something else to configure in order to place a complete call, why does the workbook not give anything about it (for all SIP Trunk tasks given, in both Volume 1 & 2).


4/ Please notice that in order to check the correct configuration, it is written the same for both "H323 Gateway" and "CCM SIP Trunk"
-> "Verification : To verify the gateway is working correctly, place a call from the PSTN phone to any BR1 on-site IP Phone, e.g. number “3123YY2001” "
But it works fine with the "H323 Gateway" task only, concerning a complete call...

Thanks for your help !

SB.

Comments

  • Hello sb.ccie,

     

    In the post you are refer "http://ieoc.com/forums/t/4986.aspx " I didn't update it  because I haven't come with a "formal" conclusion and it seems that you have made a better search from me on the SRND, only help I can give you is from what I believe and what I have realize from completing those lads.

     

    I have to note that what voicejunckie replays to me to the topic you refer is correct also I have watch ATC-VOICE and Brian Dennis mentioned the same think in the session of CCM Media Recourses.

     

    First of all you can not uncheck the box "Media Termination Point Required" this means that you play SIP early offer media with g711ulaw or g711alaw.

     

    Also you can leave the Media Resource List to none as long you don't assign the Software MTPs of CCMs in any Media Resource Group and finally in any Media Recourse List .

     

    Keep in mind that the software MTPs they can only transcode form g711ulaw to g711alw or backwards. If you want to go from g729 to g711 you have to use hardware dsp for the transcoding.

     

    So you have to place manually the codec as g711ulaw on the dial-peers voip of the SIP Cateway which points to the CCMs .And this is because the default codec on the dial-peer voip on the Cisco router is g729.If you leave it on the defaults you have to make transcode.

     

    I hope my conclusion helped you and not confuse more.

     

    Best Regards

     

    Fireman17

  • Hello Fireman17,



    Thank you for your answer: it helped a lot !


    I tried during a new lab session the solution you proposed me and yes, if you place manually the codec to g711ulaw on the voip dial-peer, it works fine. As the MTP is required (checkbox not "uncheckable"), we can write too that the CCM service "Cisco IP Voice Media Streaming Application" must be started in order to use the CCM software MTP.

    And your are correct : you can let the Media Ressource List to <none> as long as you don't assign the CCM software MTPs to any Media Resource Group and any Media Recourse List).



    Thanks for your help,
    SB.

    PS : To Internetwork Expert authors -> it could have be nice to have your own advices and comments concerning those topics as long as it seems that the "CCM Sip Trunk" Lab sems to be not fully functionnal if we only use the solution you give in the workbook, and as long as this forum is dedicated to your products.

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