Hello! This thread again. There are two logs. First between Cisco<->Cisco call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As I can see the difference is that Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a problem? Is this field mandatory? Jul 6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:28032@192.168.35.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F Remote-Party-ID: "test" <sip:30030@10.240.14.15>;party=calling;screen=no;privacy=off From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2076058060-2805141984-2507440228-2131084746 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: 1309976439 Contact: <sip:30030@10.240.14.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187 v=0 o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15 s=SIP Call c=IN IP4 10.240.14.15 t=0 0 m=audio 18720 RTP/AVP 8 c=IN IP4 10.240.14.15 a=rtpmap:8 PCMA/8000 a=ptime:20 Jul 6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Jul 6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15>;tag=CBEFC0-16E4 Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:28032@192.168.35.15:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 194 When call is between Cisco<->Freeswitch messages are Jul 6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:73435230030@192.168.3.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp Max-Forwards: 67 From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Contact: <sip:mod_sofia@192.168.1.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20 22-43-50 -0300 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 329 X-FS-Support: update_display Remote-Party-ID: "anonymous" <sip:0000000000@192.168.1.1>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0 m=audio 20360 RTP/AVP 8 0 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 19030 RTP/AVP 31 34 98 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000 Jul 6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Date: Wed, 06 Jul 2011 18:21:11 GMT Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 -- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005
http://wiki.freeswitch.org/wiki/Variable_verbose_sdp Set this to true in your dialplan before calling bridge when placing a call to the Cisco. On Thu, Jul 7, 2011 at 2:07 AM, Boris Kovalenko <boris@tagnet.ru> wrote:
Hello!
This thread again. There are two logs. First between Cisco<->Cisco call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As I can see the difference is that Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a problem? Is this field mandatory?
Jul 6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:28032@192.168.35.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F Remote-Party-ID: "test" <sip:30030@10.240.14.15>;party=calling;screen=no;privacy=off From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2076058060-2805141984-2507440228-2131084746 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: 1309976439 Contact: <sip:30030@10.240.14.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187
v=0 o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15 s=SIP Call c=IN IP4 10.240.14.15 t=0 0 m=audio 18720 RTP/AVP 8 c=IN IP4 10.240.14.15 a=rtpmap:8 PCMA/8000 a=ptime:20
Jul 6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
Jul 6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test" <sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To: <sip:28032@192.168.35.15>;tag=CBEFC0-16E4 Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:28032@192.168.35.15:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 194
When call is between Cisco<->Freeswitch messages are Jul 6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:73435230030@192.168.3.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp Max-Forwards: 67 From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Contact: <sip:mod_sofia@192.168.1.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20 22-43-50 -0300 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 329 X-FS-Support: update_display Remote-Party-ID: "anonymous" <sip:0000000000@192.168.1.1>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0
m=audio 20360 RTP/AVP 8 0 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 19030 RTP/AVP 31 34 98 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000
Jul 6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp From: "anonymous" <sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To: <sip:73435230030@192.168.3.6:5060> Date: Wed, 06 Jul 2011 18:21:11 GMT Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
-- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
Hello! I tried but.... it works only with <action application="export" data="verbose_sdp=true"/>, it this right? I thinked simple "set" is enough.
http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
Set this to true in your dialplan before calling bridge when placing a call to the Cisco.
On Thu, Jul 7, 2011 at 2:07 AM, Boris Kovalenko<boris@tagnet.ru> wrote:
Hello!
This thread again. There are two logs. First between Cisco<->Cisco call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As I can see the difference is that Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a problem? Is this field mandatory?
Jul 6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:28032@192.168.35.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F Remote-Party-ID: "test" <sip:30030@10.240.14.15>;party=calling;screen=no;privacy=off From: "test"<sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To:<sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2076058060-2805141984-2507440228-2131084746 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: 1309976439 Contact:<sip:30030@10.240.14.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187
v=0 o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15 s=SIP Call c=IN IP4 10.240.14.15 t=0 0 m=audio 18720 RTP/AVP 8 c=IN IP4 10.240.14.15 a=rtpmap:8 PCMA/8000 a=ptime:20
Jul 6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test"<sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To:<sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
Jul 6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test"<sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To:<sip:28032@192.168.35.15>;tag=CBEFC0-16E4 Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact:<sip:28032@192.168.35.15:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 194
When call is between Cisco<->Freeswitch messages are Jul 6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:73435230030@192.168.3.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp Max-Forwards: 67 From: "anonymous"<sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To:<sip:73435230030@192.168.3.6:5060> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Contact:<sip:mod_sofia@192.168.1.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20 22-43-50 -0300 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 329 X-FS-Support: update_display Remote-Party-ID: "anonymous" <sip:0000000000@192.168.1.1>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0
m=audio 20360 RTP/AVP 8 0 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 19030 RTP/AVP 31 34 98 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000
Jul 6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp From: "anonymous"<sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To:<sip:73435230030@192.168.3.6:5060> Date: Wed, 06 Jul 2011 18:21:11 GMT Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
-- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005
Export would set it on the a and b legs (unless you use nolocal). That is what you want. On Thu, Jul 7, 2011 at 10:36 AM, Boris Kovalenko <boris@tagnet.ru> wrote:
Hello!
I tried but.... it works only with <action application="export" data="verbose_sdp=true"/>, it this right? I thinked simple "set" is enough.
http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
Set this to true in your dialplan before calling bridge when placing a call to the Cisco.
On Thu, Jul 7, 2011 at 2:07 AM, Boris Kovalenko<boris@tagnet.ru> wrote:
Hello!
This thread again. There are two logs. First between Cisco<->Cisco call and second between Cisco<->FreeSwitch. Cisco<->Cisco works fine. As I can see the difference is that Cisco has a=rtpmap:8 PCMA/8000 and FreeSwitch not. May be this is a problem? Is this field mandatory?
Jul 6 18:20:39.341: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:28032@192.168.35.15:5060 SIP/2.0 Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F Remote-Party-ID: "test" <sip:30030@10.240.14.15>;party=calling;screen=no;privacy=off From: "test"<sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To:<sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Supported: 100rel,timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 2076058060-2805141984-2507440228-2131084746 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: 1309976439 Contact:<sip:30030@10.240.14.15:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 187
v=0 o=CiscoSystemsSIP-GW-UserAgent 0 703 IN IP4 10.240.14.15 s=SIP Call c=IN IP4 10.240.14.15 t=0 0 m=audio 18720 RTP/AVP 8 c=IN IP4 10.240.14.15 a=rtpmap:8 PCMA/8000 a=ptime:20
Jul 6 18:20:39.361: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test"<sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To:<sip:28032@192.168.35.15> Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
Jul 6 18:20:39.369: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.240.14.15:5060;branch=z9hG4bK797E4F From: "test"<sip:30030@10.240.14.15>;tag=3F5B1E64-2263 To:<sip:28032@192.168.35.15>;tag=CBEFC0-16E4 Date: Wed, 06 Jul 2011 18:20:39 GMT Call-ID: 7C686709-A73311E0-95798064-7F05C5CA@10.240.14.15 Timestamp: 1309976439 CSeq: 101 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact:<sip:28032@192.168.35.15:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 194
When call is between Cisco<->Freeswitch messages are Jul 6 18:21:11.966: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:73435230030@192.168.3.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp Max-Forwards: 67 From: "anonymous"<sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To:<sip:73435230030@192.168.3.6:5060> Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Contact:<sip:mod_sofia@192.168.1.1:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1c95ad9 2011-01-20 22-43-50 -0300 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY, PUBLISH, SUBSCRIBE Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 329 X-FS-Support: update_display Remote-Party-ID: "anonymous" <sip:0000000000@192.168.1.1>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1309956111 1309956112 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 t=0 0
m=audio 20360 RTP/AVP 8 0 18 101 13 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 19030 RTP/AVP 31 34 98 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:98 H264/90000
Jul 6 18:21:11.986: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.1;rport;branch=z9hG4bKe5caHt585e7Kp From: "anonymous"<sip:0000000000@192.168.1.1>;tag=QFKH52Z9NaN2c To:<sip:73435230030@192.168.3.6:5060> Date: Wed, 06 Jul 2011 18:21:11 GMT Call-ID: 91a35a9f-229f-122f-fa9c-002354cb08ac CSeq: 14652171 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0
-- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
participants (2)
-
Boris Kovalenko
-
Kristian Kielhofner