"CCM SIP Trunk" task issue
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.
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
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 !