Hi All,
During testing of a new installed Cisco IP telephony system we found that calls from analog phones, which is transfered to PSTN from local IP phone, fail.
Network topology is depicted below:
CUCM -- h.323 -- Voice GW (2821) -- SIP (PSTN)
| |
IP phone E1
|
Old telephony system -- analog phone
I collected few debug messages from gateway (called number - 1111111, calling number - 0502251822):
INVITE sip:0502251822@10.1.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK5D4013E4
Remote-Party-ID: <sip:1111111@10.1.10.12>;party=calling;screen=yes;privacy=off
From: <sip:1111111@10.1.10.10>;tag=10C604C8-1CCC
To: <sip:0502251822@10.1.10.10>
Date: Thu, 21 Jul 2011 11:49:42 GMT
Call-ID: 5B38FFDC-B2C611E0-B38BDCB5-F96223DB@10.1.10.12
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0008658550-2433778146-1967823105-3232240967
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1311248982
Contact: <sip:1111111@10.1.10.12:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 241
v=0
o=CiscoSystemsSIP-GW-UserAgent 7235 6646 IN IP4 10.1.10.12
s=SIP Call
c=IN IP4 10.1.10.12
t=0 0
m=audio 17566 RTP/AVP 0 101
c=IN IP4 10.1.10.12
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
*Jul 21 11:49:42.520: //84424/00841E76754A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK5D4013E4
From: <sip:1111111@10.1.10.10>;tag=10C604C8-1CCC
To: <sip:0502251822@10.1.10.10>;tag=1c699106095
Call-ID: 5B38FFDC-B2C611E0-B38BDCB5-F96223DB@10.1.10.12
CSeq: 101 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.019.006
Content-Length: 0
*Jul 21 11:49:42.544: //84424/00841E76754A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK5D4013E4
From: <sip:1111111@10.1.10.10>;tag=10C604C8-1CCC
To: <sip:0502251822@10.1.10.10>;tag=1c699106095
Call-ID: 5B38FFDC-B2C611E0-B38BDCB5-F96223DB@10.1.10.12
CSeq: 101 INVITE
Contact: <sip:1039@10.1.10.10>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.019.006
Content-Type: application/sdp
Content-Length: 258
v=0
o=AudiocodesGW 699156553 699156270 IN IP4 10.1.10.11
s=Phone-Call
c=IN IP4 10.1.10.11
t=0 0
m=audio 6390 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6391 IN IP4 10.1.10.11
*Jul 21 11:49:42.588: //84424/00841E76754A/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:0502251822@10.1.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK5D4013E4
From: <sip:1111111@10.1.10.10>;tag=10C604C8-1CCC
To: <sip:0502251822@10.1.10.10>
Date: Thu, 21 Jul 2011 11:49:42 GMT
Call-ID: 5B38FFDC-B2C611E0-B38BDCB5-F96223DB@10.1.10.12
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1311248982
Reason: Q.850;cause=47
Content-Length: 0
*Jul 21 11:49:42.632: //84424/00841E76754A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK5D4013E4
From: <sip:1111111@10.1.10.10>;tag=10C604C8-1CCC
To: <sip:0502251822@10.1.10.10>;tag=1c699106095
Call-ID: 5B38FFDC-B2C611E0-B38BDCB5-F96223DB@10.1.10.12
CSeq: 101 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.019.006
Reason: SIP ;cause=487 ;text="487 Request Terminated"
Content-Length: 0
*Jul 21 11:49:42.636: //84424/00841E76754A/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK5D4013E4
From: <sip:1111111@10.1.10.10>;tag=10C604C8-1CCC
To: <sip:0502251822@10.1.10.10>;tag=1c699106095
Call-ID: 5B38FFDC-B2C611E0-B38BDCB5-F96223DB@10.1.10.12
CSeq: 101 CANCEL
Contact: <sip:1039@10.1.10.10>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.019.006
Content-Length: 0
From this debug I saw that our system sent Cansel with Reason: Q.850; cause=47 (indicates a "resource unavailable" event. and typical scenarios include: out of memory; internal access to the TCP socket is unavailable. I found this parameters through the link http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/voice_troubleshooting/old/vts_appa.html#wp1007443).
When call received from local IP phone is transfered by local IP phone to PSTN it succeeded:
Sent:
INVITE sip:0502251822@10.1.10.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK57EB19F5
Remote-Party-ID: <sip:1111111@10.1.10.12>;party=calling;screen=yes;privacy=off
From: <sip:1111111@10.1.10.10>;tag=103D1604-75
To: <sip:0502251822@10.1.10.10>
Date: Thu, 21 Jul 2011 09:20:08 GMT
Call-ID: 764AEC13-B2B111E0-9240DCB5-F96223DB@10.1.10.12
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0013185681-2213376482-1682312705-3232289737
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1311240008
Contact: <sip:1111111@10.1.10.12:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 241
v=0
o=CiscoSystemsSIP-GW-UserAgent 5587 1970 IN IP4 10.1.10.12
s=SIP Call
c=IN IP4 10.1.10.12
t=0 0
m=audio 17426 RTP/AVP 0 101
c=IN IP4 10.1.10.12
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
*Jul 21 09:20:08.493: //79402/00C932916446/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK57EB19F5
From: <sip:1111111@10.1.10.10>;tag=103D1604-75
To: <sip:0502251822@10.1.10.10>;tag=1c804216384
Call-ID: 764AEC13-B2B111E0-9240DCB5-F96223DB@10.1.10.12
CSeq: 101 INVITE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.019.006
Content-Length: 0
*Jul 21 09:20:08.541: //79402/00C932916446/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK57EB19F5
From: <sip:1111111@10.1.10.10>;tag=103D1604-75
To: <sip:0502251822@10.1.10.10>;tag=1c804216384
Call-ID: 764AEC13-B2B111E0-9240DCB5-F96223DB@10.1.10.12
CSeq: 101 INVITE
Contact: <sip:1051@10.1.10.10>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.019.006
Content-Type: appli
C2821GW2#cation/sdp
Content-Length: 258
v=0
o=AudiocodesGW 804257236 804256952 IN IP4 10.1.10.11
s=Phone-Call
c=IN IP4 10.1.10.11
t=0 0
m=audio 6510 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6511 IN IP4 10.1.10.11
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.10.12:5060;branch=z9hG4bK57EB19F5
From: <sip:1111111@10.1.10.10>;tag=103D1604-75
To: <sip:0502251822@10.1.10.10>;tag=1c804216384
Call-ID: 764AEC13-B2B111E0-9240DCB5-F96223DB@10.1.10.12
CSeq: 101 INVITE
Contact: <sip:1051@10.1.10.10>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 1000/v.5.20A.019.006
Content-Type: application/sd
C2821GW2#p
Content-Length: 258
v=0
o=AudiocodesGW 804257236 804256952 IN IP4 10.1.10.11
s=Phone-Call
c=IN IP4 10.1.10.11
t=0 0
m=audio 6510 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=rtcp:6511 IN IP4 10.1.10.11
I will be grateful for any suggestions and help in resolving this issue.
Thank you!
Kind regards,
Lidiia
