SIP Trunk

[fgasimzade]

I am trying to configure SIP trunk between Cisco Unified Communication Server and 1861 router with SRST.

When I try to initiate a call through this trunk, I hear only beeps and call does not succeed. However, in debug voice dialpeer all I can see something going on, dont know exactly what it means, but I suppose SIP trunk does not go up.

Take a look at my SIP config on the router and let me know please what I am missing:

voice service voip

allow-connections sip to sip

no supplementary-service sip moved-temporarily

supplementary-service ringback h225-info

fax protocol cisco

h323

call start slow

sip

no call service stop

----

dial-peer voice 3 voip

destination-pattern ....

session protocol sipv2

session target ipv4:192.168.16.6

dtmf-relay rtp-nte

codec g711alaw

no vad

Thank you in advance

[Nicholas Matthews]

debug ccsip messages may be more informative of your problem.

-nick

[fgasimzade]

Thank you.

Config is ok?

[Nicholas Matthews]

looks like it could work. context is more important than configuration.

-nick

[fgasimzade]

Thank you Nick! I managed to call to remote peer through SIP trunk, but I soon as I pick up the phone I hear beeps and get this debug messages:

Sep 4 09:54:32.179: //17/C76106C4803A/SIP/Error/sipSPIAddCiscoGcid: Fatal Error

in parsing CCB/Msg

SIP: Attribute mid, level 1 instance 1 not found.

Sep 4 09:54:33.403: //18/C76106C4803A/SIP/Error/sipSPIDoAudioNegotiation: Media

negotiation failed for m-line 1

Sep 4 09:54:33.403: //18/C76106C4803A/SIP/Error/sipSPIDoMediaNegotiation:

no valid fax or audio streams

Sep 4 09:54:33.403: //18/C76106C4803A/SIP/Error/ccsip_api_call_cut_progress: Me

diaNegotiation Failure - Send Cancel

Sep 4 09:54:33.403: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG: No Inboun

d Container Created !!!

Sep 4 09:54:33.403: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inboun

d Container Created !!!

Sep 4 09:54:33.419: //17/C76106C4803A/SIP/Error/act_sentalert_disconnect: Disco

nnect cause from ccapi :65 and sip_disc_cause: 501

Was unable to find any solutions on the net.

[fgasimzade]

Also, from show call history voice I get this message

DisconnectCause=41

DisconnectText=bearer capability not implemented (65)

[fgasimzade]

Issue is solved, it was a codec mismatch, thank you!