Call Diversion/CF (302 Moved Temporarily) not working
Hi, My setup looks like this: User A (1001), User B (1002) and User C (1003) are registered to 13.13.13.66 via OpenSIPs Proxy 222.222.222.222. User B activated unconditional call forwarding on his phone to User C. Now User A (or any external caller) calls User B. Until last week User A would then be able to talk to User C - the expected behaviour. On Saturday, I updated the freeswitch from version 9fe08675a1 to d1c3f910a6 to get and test some of Steve's fax changes. Since this upgrade, Call-Forwarding / Call Diversion is broken on a level, that I don't understand. User A: INVITE 1002 to B's phone => User B: 302 Moved Temporarily (Contact: <sip:1003@sip.isp.net;user=phone>) => FS: ACK => FS: INVITE 1003 to *B*'s phone! => User B: 404 Not Found => FS: ACK => User A gets signalled the 404 Not Found. packet trace below. I'm aware of http://jira.freeswitch.org/browse/FS-724 http://jira.freeswitch.org/browse/FS-821 but those didn't help me solve my issue: internal.xml: <param name="apply-nat-acl" value="nat.auto"/> internal.xml: <param name="aggressive-nat-detection" value="false"/> I added aggressive-nat-detection=false yesterday, before it wasn't in the config at all. I also tried removing the nat.auto acl temporarily, but that didn't help either. Yesterday afternoon I updated to git c3de9637af, did several bootstrappings, cleans and makes... no change. So today I tried to downgrade to 9fe08675a1, the version I had running until Friday and that was working fine... well, it doesn't now. So I can't even _prove_ it was working before :( Can somebody please point me in the right direction? Any hint in the right direction is much appreciated! Thanks! ====================================================================================== SIP Trace (tcpdump): 16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....[ .>..;^... .......D.fINVITE sip:1002@137.137.137.245:7390;line=ydwzxzpg SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 442 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 teleph 16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp E....[..>.4.^... ..one-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000 16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623 E.....@.@. . ..^........w8$SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:1003@sip.isp.net;user=phone> Diversion: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;reason="unconditional" Content-Length: 0 16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358 .^....@.>. .......n.fACK sip:1002@137.137.137.245:7390;line=ydwzxzpg SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0 16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....\ .>..:^... .......ZgIINVITE sip:1003@137.137.137.245:7390;user=phone SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 466 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 telephone-e 16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp E....\..>.4.^... ..vent/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000 16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551 E..C..@.@..7 ..^......../..SIP/2.0 404 Not Found Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Length: 0 16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341 E..q..@.>...^... .......].mACK sip:1003@137.137.137.245:7390;user=phone SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0 ======================================================================================
Hi guys, so really nobody got an idea about that one? It's driving me nuts ;) _Can_ this be some kind of dialplan fuckup? Regards, Johannes -- Johannes Jakob Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, 7. August 2012 at 18:03, Johannes Jakob wrote:
Hi,
My setup looks like this:
User A (1001), User B (1002) and User C (1003) are registered to 13.13.13.66 via OpenSIPs Proxy 222.222.222.222. User B activated unconditional call forwarding on his phone to User C. Now User A (or any external caller) calls User B.
Until last week User A would then be able to talk to User C - the expected behaviour. On Saturday, I updated the freeswitch from version 9fe08675a1 to d1c3f910a6 to get and test some of Steve's fax changes. Since this upgrade, Call-Forwarding / Call Diversion is broken on a level, that I don't understand.
User A: INVITE 1002 to B's phone => User B: 302 Moved Temporarily (Contact: <sip:1003@sip.isp.net (mailto:1003@sip.isp.net);user=phone>) => FS: ACK => FS: INVITE 1003 to *B*'s phone! => User B: 404 Not Found => FS: ACK => User A gets signalled the 404 Not Found.
packet trace below.
I'm aware of http://jira.freeswitch.org/browse/FS-724 http://jira.freeswitch.org/browse/FS-821
but those didn't help me solve my issue:
internal.xml: <param name="apply-nat-acl" value="nat.auto"/> internal.xml: <param name="aggressive-nat-detection" value="false"/>
I added aggressive-nat-detection=false yesterday, before it wasn't in the config at all. I also tried removing the nat.auto acl temporarily, but that didn't help either.
Yesterday afternoon I updated to git c3de9637af, did several bootstrappings, cleans and makes... no change.
So today I tried to downgrade to 9fe08675a1, the version I had running until Friday and that was working fine... well, it doesn't now. So I can't even _prove_ it was working before :(
Can somebody please point me in the right direction?
Any hint in the right direction is much appreciated!
Thanks!
======================================================================================
SIP Trace (tcpdump):
16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....[ .>..;^... .......D.fINVITE sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66 (mailto:mod_sofia@13.13.13.66):5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 442 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 teleph 16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp E....[..>.4.^... ..one-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000
16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623 E.....@.@. . ..^........w8$SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:1003@sip.isp.net (mailto:1003@sip.isp.net);user=phone> Diversion: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg>;reason="unconditional" Content-Length: 0
16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358 .^....@.>. .......n.fACK sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=3r8K7r5Hppe6N Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0
16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....\ .>..:^... .......ZgIINVITE sip:1003@137.137.137.245 (mailto:1003@137.137.137.245):7390;user=phone SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66 (mailto:mod_sofia@13.13.13.66):5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 466 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 telephone-e 16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp E....\..>.4.^... ..vent/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000
16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551 E..C..@.@..7 ..^......../..SIP/2.0 404 Not Found Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Length: 0
16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341 E..q..@.>...^... .......].mACK sip:1003@137.137.137.245 (mailto:1003@137.137.137.245):7390;user=phone SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=411c9KpNKZ4rH Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0
======================================================================================
Its been said a million times on this ML for bugs to be posted to jira, further read the wiki on using git bisect to track down the exact revision it broke in, that will greatly increase the chance of the bug being taken care of asap. ~mitch On Fri, Aug 10, 2012 at 1:43 AM, Johannes Jakob <lists.jj@googlemail.com> wrote:
Hi guys,
so really nobody got an idea about that one? It's driving me nuts ;)
_Can_ this be some kind of dialplan fuckup?
Regards,
Johannes
-- Johannes Jakob Sent with Sparrow (http://www.sparrowmailapp.com/?sig)
On Tuesday, 7. August 2012 at 18:03, Johannes Jakob wrote:
Hi,
My setup looks like this:
User A (1001), User B (1002) and User C (1003) are registered to 13.13.13.66 via OpenSIPs Proxy 222.222.222.222. User B activated unconditional call forwarding on his phone to User C. Now User A (or any external caller) calls User B.
Until last week User A would then be able to talk to User C - the expected behaviour. On Saturday, I updated the freeswitch from version 9fe08675a1 to d1c3f910a6 to get and test some of Steve's fax changes. Since this upgrade, Call-Forwarding / Call Diversion is broken on a level, that I don't understand.
User A: INVITE 1002 to B's phone => User B: 302 Moved Temporarily (Contact: <sip:1003@sip.isp.net (mailto:1003@sip.isp.net);user=phone>) => FS: ACK => FS: INVITE 1003 to *B*'s phone! => User B: 404 Not Found => FS: ACK => User A gets signalled the 404 Not Found.
packet trace below.
I'm aware of http://jira.freeswitch.org/browse/FS-724 http://jira.freeswitch.org/browse/FS-821
but those didn't help me solve my issue:
internal.xml: <param name="apply-nat-acl" value="nat.auto"/> internal.xml: <param name="aggressive-nat-detection" value="false"/>
I added aggressive-nat-detection=false yesterday, before it wasn't in the config at all. I also tried removing the nat.auto acl temporarily, but that didn't help either.
Yesterday afternoon I updated to git c3de9637af, did several bootstrappings, cleans and makes... no change.
So today I tried to downgrade to 9fe08675a1, the version I had running until Friday and that was working fine... well, it doesn't now. So I can't even _prove_ it was working before :(
Can somebody please point me in the right direction?
Any hint in the right direction is much appreciated!
Thanks!
======================================================================================
SIP Trace (tcpdump):
16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....[ .>..;^... .......D.fINVITE sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66 (mailto:mod_sofia@13.13.13.66):5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 442 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 teleph 16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp E....[..>.4.^... ..one-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000
16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623 E.....@.@. . ..^........w8$SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:1003@sip.isp.net (mailto:1003@sip.isp.net);user=phone> Diversion: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg>;reason="unconditional" Content-Length: 0
16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358 .^....@.>. .......n.fACK sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=3r8K7r5Hppe6N Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg>;tag=mzey6xfxhz CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0
16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....\ .>..:^... .......ZgIINVITE sip:1003@137.137.137.245 (mailto:1003@137.137.137.245):7390;user=phone SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66 (mailto:mod_sofia@13.13.13.66):5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 466 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 telephone-e 16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp E....\..>.4.^... ..vent/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000
16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551 E..C..@.@..7 ..^......../..SIP/2.0 404 Not Found Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Length: 0
16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341 E..q..@.>...^... .......].mACK sip:1003@137.137.137.245 (mailto:1003@137.137.137.245):7390;user=phone SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 From: "User A" <sip:1001@sip.isp.net (mailto:1001@sip.isp.net)>;tag=411c9KpNKZ4rH Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245 (mailto:1002@137.137.137.245):7390;line=ydwzxzpg> CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0
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participants (2)
-
Johannes Jakob
-
Mitch Capper