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Re: [Freeswitch-users] Cisco 7960 has problem with music on hold
by Gary Chen 04 May '10

04 May '10

Here is the detail: Both Cisco 7960 loaded with SIP firmware: P0S3-08-9-00 They both uses static IP address. I am running Freeswitch 1.0.6 The following is the SIP trace: (I altered IP addresses) U 222.111.000.144:50745 -> 222.111.000.177:5060 INVITE sip:1004@222.111.000.177;user=phone SIP/2.0..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008@222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004@222.111.000.177;user=phone>..Cal l-ID: 00082166-efcb0019-398a09d5-5e799be0@222.111.000.144..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:11 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1008@222.111.000.144:5060;transport=udp>..Expires: 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub ..Content-Length: 282..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIP UA 11004 0 IN IP4 222.111.000.144..s=SIP Call..t=0 0..m=audio 21654 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.144..a=rtpm ap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=f mtp:101 0-15..a=sendrecv.. # U 222.111.000.177:5060 -> 222.111.000.144:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008@222.111.000.17 7>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004@222.111.000.177;user=phone>..Call-ID: 00082166-efcb0019-398a09d5 -5e799be0@222.111.000.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Content-Length: 0.... # U 222.111.000.177:5060 -> 222.111.000.102:5060 INVITE sip:1004@222.111.000.102:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm 350B9r..Max-Forwards: 68..From: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..To: <sip:1004@222.111.000.102:5060 ;transport=udp>..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 130369205 INVITE..Contact: <sip:mod_sofia@222.111. 120.177:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, U PDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. .Content-Type: application/sdp..Content-Disposition: session..Content-Length: 433..X-FS-Support: update_display..Remote- Party-ID: "Line2" <sip:1008@222.111.000.177>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1272956499 127295 6500 IN IP4 222.111.000.177..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 115 107 9 8 3 101 13 ..a=rtpmap:0 PCMU/8000..a=rtpmap:115 G7221/32000..a=fmtp:115 bitrate=48000..a=rtpmap:107 G7221/16000..a=fmtp:107 bitrate =32000..a=rtpmap:9 G722/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0 -16..a=rtpmap:13 CN/8000..a=ptime:20.. # U 222.111.000.102:51984 -> 222.111.000.177:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008@222.111. 120.177>;tag=9BNeN21mSySQN..To: <sip:1004@222.111.000.102:5060;transport=udp>..Call-ID: ff905261-d218-122d-9ab7-6d55cd3b b1e9..Date: Tue, 04 May 2010 12:17:12 GMT..CSeq: 130369205 INVITE..Server: Cisco-CP7960G/8.0..Contact: <sip:1001@222.111 .000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Content-Length: 0.... # U 222.111.000.102:51985 -> 222.111.000.177:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008@222.111 .000.177>;tag=9BNeN21mSySQN..To: <sip:1004@222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Ca ll-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:12 GMT..CSeq: 130369205 INVITE..Server: Cisco- CP7960G/8.0..Contact: <sip:1004@222.111.000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,R EGISTER,UPDATE..Content-Length: 0.... # U 222.111.000.177:5060 -> 222.111.000.144:5060 SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008@204. 126.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004@222.111.000.177;user=phone>;tag=82UNK7gHvN34S..Call-ID : 00082166-efcb0019-398a09d5-5e799be0@222.111.000.144..CSeq: 101 INVITE..Contact: <sip:1004@222.111.000.177:5060;transpo rt=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, mess age-summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 253..Remote-Party-ID: " 1004" <sip:1004@222.111.000.177>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1272949011 1272949012 IN IP4 2 04.126.000.177..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 26072 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpm ap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20.. # U 222.111.000.102:51986 -> 222.111.000.177:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008@222.111.000. 177>;tag=9BNeN21mSySQN..To: <sip:1004@222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Call-ID : ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:13 GMT..CSeq: 130369205 INVITE..Server: Cisco-CP796 0G/8.0..Contact: <sip:1004@222.111.000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGIST ER,UPDATE..Supported: replaces,join,norefersub..Content-Length: 210..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 5247 0 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP 0 101..c=IN IP4 222.111.000.102..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. # U 222.111.000.177:5060 -> 222.111.000.102:5060 ACK sip:1004@222.111.000.102:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKcepB2m5629 1Um..Max-Forwards: 70..From: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..To: <sip:1004@222.111.000.102:5060;tr ansport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 130369205 ACK.. Contact: <sip:mod_sofia@222.111.000.177:5060>..Content-Length: 0.... # U 222.111.000.177:5060 -> 222.111.000.144:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008@222.111.000.177>;t ag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004@222.111.000.177;user=phone>;tag=82UNK7gHvN34S..Call-ID: 00082166-efc b0019-398a09d5-5e799be0@222.111.000.144..CSeq: 101 INVITE..Contact: <sip:1004@222.111.000.177:5060;transport=udp>..User- Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTE R, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dial og, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Content-Type: appli cation/sdp..Content-Disposition: session..Content-Length: 253..Remote-Party-ID: "1004" <sip:1004@222.111.000.177>;party= calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1272949011 1272949012 IN IP4 222.111.000.177..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 26072 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20.. # U 222.111.000.144:50746 -> 222.111.000.177:5060 ACK sip:1004@222.111.000.177:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK27a3b7e3..F rom: "Line2" <sip:1008@222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004@222.111.000.177;user=phone> ;tag=82UNK7gHvN34S..Call-ID: 00082166-efcb0019-398a09d5-5e799be0@222.111.000.144..Max-Forwards: 70..Date: Tue, 04 May 20 10 12:17:14 GMT..CSeq: 101 ACK..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0.... # U 222.111.000.102:51018 -> 222.111.000.177:5060 INVITE sip:mod_sofia@222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: < sip:1004@222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.17 7>;tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:15 G MT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1004@222.111.000.102:5060;transport=udp>..Accept: app lication/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Co ntent-Length: 281..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 5 247 1 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.102..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-15..a=sendonly.. # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004@222.111.000.102:5060;t ransport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Content-Len gth: 0.... # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004@222.111.000.102:5060;trans port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90 5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia@222.111.000.177:5060>..User-Agent: FreeSWITC H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17 7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20.. # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004@222.111.000.102:5060;trans port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90 5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia@222.111.000.177:5060>..User-Agent: FreeSWITC H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17 7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20.. # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004@222.111.000.102:5060;trans port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90 5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia@222.111.000.177:5060>..User-Agent: FreeSWITC H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17 7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20.. # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004@222.111.000.102:5060;trans port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90 5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia@222.111.000.177:5060>..User-Agent: FreeSWITC H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17 7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20.. # U 222.111.000.102:51018 -> 222.111.000.177:5060 INVITE sip:mod_sofia@222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: < sip:1004@222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.17 7>;tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:22 G MT..CSeq: 102 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1004@222.111.000.102:5060;transport=udp>..Accept: app lication/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Co ntent-Length: 281..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 5 247 2 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.102..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-15..a=sendrecv.. # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 500 Overlapping Requests..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <sip:1004@222.111. 120.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSy SQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 102 INVITE..Retry-After: 2 (Overlapping Requests)..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UP DATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Length: 0.... # U 222.111.000.102:51987 -> 222.111.000.177:5060 ACK sip:mod_sofia@222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <sip :1004@222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>; tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:23 GMT..CSeq: 102 ACK..Co ntent-Length: 0.... # U 222.111.000.102:51018 -> 222.111.000.177:5060 BYE sip:mod_sofia@222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK7102f98b..From: <sip :1004@222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>; tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:23 GMT. .CSeq: 103 BYE..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0.... # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK7102f98b..From: <sip:1004@222.111.000.102:5060;trans port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90 5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 103 BYE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK , BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, preconditi on, path, replaces..Content-Length: 0.... # U 222.111.000.177:5060 -> 222.111.000.144:5060 BYE sip:1008@222.111.000.144:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKDQF43Fpa0j reg..Max-Forwards: 70..From: <sip:1004@222.111.000.177;user=phone>;tag=82UNK7gHvN34S..To: "Line2" <sip:1008@222.111.000. 177>;tag=00082166efcb01396b803ac6-7e5ae7dc..Call-ID: 00082166-efcb0019-398a09d5-5e799be0@222.111.000.144..CSeq: 13036921 1 BYE..Contact: <sip:1004@222.111.000.177:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allo w: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: tim er, precondition, path, replaces..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0.... # U 222.111.000.144:50747 -> 222.111.000.177:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKDQF43Fpa0jreg..From: <sip:1004@222.111.000.177;user =phone>;tag=82UNK7gHvN34S..To: "Line2" <sip:1008@222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..Call-ID: 000821 66-efcb0019-398a09d5-5e799be0@222.111.000.144..Date: Tue, 04 May 2010 12:17:24 GMT..CSeq: 130369211 BYE..Server: Cisco-C P7960G/8.0..Content-Length: 0.... # U 222.111.000.177:5060 -> 222.111.000.102:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004@222.111.000.102:5060;trans port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90 5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia@222.111.000.177:5060>..User-Agent: FreeSWITC H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17 7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20.. # U 222.111.000.102:51988 -> 222.111.000.177:5060 ACK sip:mod_sofia@222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c87fac1..From: <sip :1004@222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008@222.111.000.177>; tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:16 GMT. .CSeq: 101 ACK..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0.... exit 25 received, 0 dropped

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wrong ptimes and second SDP!, cough cough Thomson ST2030
by Alberto Escudero 04 May '10

04 May '10

I have a Thomson ST2030 VoIP phone that seams to be sending wrong ptimes. The SDP says 20 ms when it should be sending 10 ms. Freeswitch *cough cough* algorithm is able to set the soft timer for 10 ms... but when another SDP arrives (after approx. 30-40 seconds) the ptime is set back again to 20 ms. I am using 1.0.5 and it seems a bug in Thomson SIP stack. Any work around? -- Stopping junk mailers is good for the environment

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Re: [Freeswitch-users] Aastra and SCA
by Brian West 04 May '10

04 May '10

Give up their exists a bug that prevents it from working on any aastra. Polycom and Cisco SPA work flawless. /b Sent from my iPad On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly(a)molcs.org> wrote: > Hi, > > > > I’ve tried to patiently figure this out by reading the wiki and this list. Unfortunately, I’ve been unable to get it right. > > > > I have 2 Aastra 6731is and a 51i and trying to get SCA working. I’m experiencing problems similar to the Cisco thread from last month – outgoing calls implement SCA well. Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones. I am not attempting to use any DNS. > > > > My configuration has all three phones plus FS on a local LAN. Nothing too fancy. Each line is configured for Broadsoft SCA and SCA bridging is enabled globally for the phones. > > > > Trying to update/replace an old phone at my church. > > > > Any help would be greatly appreciated. > > > > Thanks, Mark > > Mark.maly(a)molcs.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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Re: [Freeswitch-users] Aastra and SCA
by Michael Jerris 04 May '10

04 May '10

Aastra has a broken SCA implementation. We have been working with them to resolve it but as of yet they have failed to deliver a working firmware for their phones. Mike On Apr 30, 2010, at 10:07 PM, Mark Maly wrote: > Hi, > > I’ve tried to patiently figure this out by reading the wiki and this list. Unfortunately, I’ve been unable to get it right. > > I have 2 Aastra 6731is and a 51i and trying to get SCA working. I’m experiencing problems similar to the Cisco thread from last month – outgoing calls implement SCA well. Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones. I am not attempting to use any DNS. > > My configuration has all three phones plus FS on a local LAN. Nothing too fancy. Each line is configured for Broadsoft SCA and SCA bridging is enabled globally for the phones. > > Trying to update/replace an old phone at my church.

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Trouble With Transfer and Async Event Socket
by Joe Crivello 03 May '10

03 May '10

Hello All, I am currently working on a conference bridge based on FreeSWITCH that allows some conference members to dial out to a prompted number. Currently my implementation binds * in conference.xml to the "transfer" action, leading to a dial plan extension that runs an async full outgoing event socket. My application serving the event socket does the following: 1) Verifies the user is allowed to dial out, and writes a record to the database. 2) Retrieves the number to be dialed with 'play_and_get_digits' 3) Bridges the call to the dialed number 4) Waits for the a-leg to press star 5) Uses 'transfer' to transfer both legs back to the conference Everything works up to step #5, which does nothing. There is no NOTICE outputted to the console indicating that a transfer is taking place and neither leg is rejoined to the conference. I get a normal event back after the transfer sendmsg: Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK If I subscribe to all events before running the transfer and receive events in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events after the transfer. I am using ESL to communicate with FreeSWITCH, and prior to running any of steps #1-5 I set the event lock to true. The line from the dial plan extension that calls my application follows: <action application="socket" data="172.18.11.1:8001 async full" /> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS. Thanks! Joe Crivello

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Cisco 7960 has problem with music on hold
by Gary Chen 03 May '10

03 May '10

I just installed FS and registered two cisco 7960 with default configuration file. They can talk to each other. After connecting both phones, I can put one on hold with music, but I can not get the connection back by pressing resume softkey button. Once I pressing resume button, the connection was hangup. I can see the cisco 7960 sent INVITE to freeswith once I press resume button but FS sent back 'SIP/2.0 500 Overlapping Requests ' message. This cause cisco 7960 sent out BYE message. Any idea? Gary

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Anyone have analog Fax working through FS and Sangoma cards?
by Robert Hadley 03 May '10

03 May '10

Does anyone have analog Fax working through Freeswitch and Sangoma A101 & A200/FXS cards? If so, would you share what configuration and dialplan settings need to be made? I have a problem receiving Faxes, which only work 80% of the time and often take 10-12 minutes per Fax, versus 2 minutes when using the Fax machine on analog POTS line. My setup: SendingFax --- {PSTN} --- PRI --- A101 --- FS --- A200/FXS --- MyFax Pastebin to dialplan, wanpipe conf, vars.xml, internal.xml: http://pastebin.freeswitch.org/12884 Any help would be greatly appreciated. Thanks, Robert

2 1

Trouble With 'transfer' DP After a 'bridge' on Async Event Socket
by Joseph Crivello 03 May '10

03 May '10

Hello All, I am currently working on a conference bridge based on FreeSWITCH that allows some conference members to dial out to a prompted number. Currently my implementation binds * in conference.xml to the "transfer" action, leading to a dial plan extension that runs an async full outgoing event socket. My application serving the event socket does the following: 1) Verifies the user is allowed to dial out, and writes a record to the database. 2) Retrieves the number to be dialed with 'play_and_get_digits' 3) Bridges the call to the dialed number 4) Waits for the a-leg to press star 5) Uses 'transfer' to transfer both legs back to the conference Everything works up to step #5, which does nothing. There is no NOTICE outputted to the console indicating that a transfer is taking place and neither leg is rejoined to the conference. I get a normal event back after the transfer sendmsg: Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK If I subscribe to all events before running the transfer and receive events in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events after the transfer. I am using ESL to communicate with FreeSWITCH, and prior to running any of steps #1-5 I set the event lock to true. The line from the dial plan extension that calls my application follows: <action application="socket" data="172.18.11.1:8001 async full" /> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS. Thanks! Joe Crivello

3 2

Need help on realtime integration Freeswitch with Opensips.
by hung nguyen 03 May '10

03 May '10

Hi list. I have deployed cluster FS with share DB - using ODBC in core. I think it is great if we can deploy realtime integration Freeswitch with Opensips. With this, Opensips will act as register,proxy server and load balancer in front of FS boxs. Anybody have same ideal and done this. Tks.

2 1

Strange errors from sqlite DB
by Peter Olsson 03 May '10

03 May '10

I've had an instance of FS running on a box for a couple weeks (since I last touched it), it's mostly used for trunking between some lab PBX'es and conferencing, and 2-3 Polycom's also register to it. It was compiled at april 14 from current git at that time (I can look up the exact revision if needed). Today, after a restart, I suddenly received these errors. I've never seen them before, so I was quite surprised :) Has anyone seen these before? I shut down FS after the error, removed all db-files and restarted again. The error dissapeared, but is this something I should worry about? I've been using FS (very lightly - but still..) for over a year, and this was the first time I noticed it. I found on the list that someone had a similar problem before, but I couldn't find a real explanation to it. I'm running on Windows 2003. ---- 2010-05-03 20:29:21.032501 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence] insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('6e5df620-d388-4f93-b0b0-8da4cf880ef1', 1272911421, '(NULL)', 'internal', 0) 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:722 SQL ERR: [select nonce,last_nc from sip_authentication where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' and last_nc < 1] library routine called out of sequence 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence] update sip_authentication set expires='1272911421',last_nc=1 where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence] Regards, Peter

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