Re: [Freeswitch-users] SIP Re-invite
Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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
Re: [Freeswitch-users] SIP Re-invite
I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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 ___ 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
Re: [Freeswitch-users] SIP Re-invite
Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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 ___ 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
Re: [Freeswitch-users] SIP Re-invite
Mike, There are 2 traces in there. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 in ngrep portion, but it did not show in FS siptrace debug. Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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 ___ 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
Re: [Freeswitch-users] SIP Re-invite
Thank you Mike for your suggestion on IRC. We did what you recommend and found out it's the iptables issue that we thought it was not there at the beginning since we saw the first 2 invites from the far end fine, but somehow it has something to do with the 3rd invite. I did close the Jira that I thought it was a bug. Thank you again for the community and your support. Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, December 18, 2009 9:37:16 AM Subject: Re: [Freeswitch-users] SIP Re-invite I read through the trace, can you clarify where the missing invite is? I think I see everything in the sofia trace. Mike On Dec 18, 2009, at 3:10 AM, DJB wrote: Mike, My latest traces that I captured were done within the FS box: http://pastebin.freeswitch.org/11541 Thank you, Dorn B. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 8:03:46 AM Subject: Re: [Freeswitch-users] SIP Re-invite are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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 ___ 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
Re: [Freeswitch-users] SIP Re-invite
if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. ___ 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
Re: [Freeswitch-users] SIP Re-invite
Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
Is the packet capture running on the FS box itself? On Thu, Dec 17, 2009 at 9:36 AM, Michael Jerris m...@jerris.com wrote: if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. -- *From:* Anthony Minessale anthony.miness...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Wed, December 16, 2009 3:42:48 PM *Subject:* Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
are you doing this trace from the freeswitch box itself? Mike On Dec 17, 2009, at 10:48 AM, DJB wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. ___ 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
Re: [Freeswitch-users] SIP Re-invite
I'd be suspicious of: (a) the CSeq going backwards from 4 to 3 between re-invites 2 and 3; (b) the branch on the Via tag changing (c) (not sure about this one) the SDP session ID and version changing for what's the same session. --Dave Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. __ From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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 ___ 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
Re: [Freeswitch-users] SIP Re-invite
The trace that I pasted on the pastebin was from our analyzer,Tektronix spectra2 that was sitting between FS and customer. I also had the FS sip trace on and compare with the trace from Spectra when I found out about the 3rd re-invite was missing from FS. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn: +19193869900 +19193869900 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
It only happened to the calls from this customer that keeps sending re-invite every 30 minutes, since their switch is expecting a reply back from those re-invite and FS did not respond back to those re-invite. Thank you. From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:36:44 AM Subject: Re: [Freeswitch-users] SIP Re-invite if you don't see it in sofia siptrace but do see it in tcpdump capture then something very ugly is going on. Either sofia has hung up completely and is not listening on that port anymore (can other calls go through?) or the packet you see in tcpdump is not really going to the right port. Can you confirm which one? Mike On Dec 16, 2009, at 6:29 PM, DJB wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. ___ 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
Re: [Freeswitch-users] SIP Re-invite
Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
I am sorry; here is the complete one: http://pastebin.freeswitch.org/11540 Thank you. From: DJB djbin...@yahoo.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 9:35:27 AM Subject: Re: [Freeswitch-users] SIP Re-invite Anthony/Michael, I finally have a complete traces of a call at http://pastebin.freeswitch.org/11539 There are 2 traces in there from within the same box. One is from freeswitch/sofia siptrace debug and the other one from ngrep for your comparison. The missing re-invite in FS is at 2009/12/17 17:25:55.207747 Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
Can you post the full packets with Ethernet, IP, UDP headers as well, or upload a pcap file? I'll add the change in 'Max-Forwards' from 70 to 69 between the two packets to my things to be suspicious of list. --Dave The trace that I pasted on the pastebin was from our analyzer,Tektronix spectra2 that was sitting between FS and customer. I also had the FS sip trace on and compare with the trace from Spectra when I found out about the 3rd re-invite was missing from FS. Thank you. __ From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, December 17, 2009 7:57:42 AM Subject: Re: [Freeswitch-users] SIP Re-invite The question was: Are you doing the packet capture on the actual FS box using tshark or tcpdump? On Thu, Dec 17, 2009 at 9:48 AM, DJB djbin...@yahoo.com wrote: Anthony, I have pasted the invite sip trace here: http://pastebin.freeswitch.org/11536 Please advise if you need further info. Thank you. __ From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn: +19193869900 +19193869900 ___ 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 -- Anthony Minessale II FreeSWITCH http
Re: [Freeswitch-users] SIP Re-invite
that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP Re-invite
FWIW, we’ve seen the same thing intermittently, haven’t had time/been able to get a solid repro to capture debug information. Call ID and tags are all matching. After the re-invite fails and the remote end sends a BYE, FS does indeed respond to the re-invite. -Michael From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of DJB Sent: Wednesday, December 16, 2009 6:00 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP Re-invite Call-ID are the same for 1st, 2nd, and 3rd INVITE. The only thing I saw difference was the Via Branch value. Would that be a problem, since 1st and 2nd INVITE was also different and was okay. Is there any other values that I should look at? Thank you. From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, December 16, 2009 3:42:48 PM Subject: Re: [Freeswitch-users] SIP Re-invite that means the invite is not matching the call dialog compare the via tags and call-id etc On Wed, Dec 16, 2009 at 5:29 PM, DJB djbin...@yahoo.commailto:djbin...@yahoo.com wrote: We have a customer that we are sending calls to off the FS and here is the issue: Call is initially setup fine and they send a first re-invite with media 0.0.0.0 to place the caller on hold. FS sends a 200 ok to this first re-invite fine They then send a second re-invite with their media IP to cut through media and the FS sends a 200 OK to this fine. At this point the call is fine 30 minutes later they send a third re-invite because according to them it is strictly for the purpose of “keep alive” per RFC 4028. This third re-invite has the exact same media IP and UDP pot information as the second re-invite does. The problem is FS does not respond to this third re-invite AT ALL. It doesn’t send a 100 trying a 200 OK nothing so this causes the call to be dropped as the other end does not recieve a response from FS. One more thing, we did not see the third re-invite in sofia siptrace, but we do see it in ethereal, which is kind of odds. We are running FreeSWITCH Version 1.0.trunk (15979) in bypass media mode. Thank you very much. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto: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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.commailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.commailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.nethttp://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.orgmailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orgmailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:+19193869900 ___ 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
Re: [Freeswitch-users] SIP re-invite / bypass_media
Thanks for that. That seems to successfully re-invite and re-route the the B leg - but does not reinvite the A leg and then immediately issues a bye on both legs. Do I have to do something to reinvite that A leg? On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale anthony.miness...@gmail.com wrote: try apiExecute(uuid_media, off + session.uuid); On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones pjinthe...@gmail.comwrote: Hi there, I was wondering whether it is possible to have FreeSwitch go into bypass_media mode on demand? For instance, leg a bridges to leg b - leg b is invited to accept the call by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute the media) after the one is pressed. Currently I am issuing the following from my js script that prompts for the 1: session.apiExecute(uuid_media,session.uuid); Not working however. Any help to get me going would be appreciated. Thanks Phillip Jones. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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 ___ 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
Re: [Freeswitch-users] SIP re-invite / bypass_media
Thanks for responding and for your help.
The xml and confirm.js are attached below. Basically trying to bypass_media
after the leg B presses 1 to accept the call. I tried,
using bypass_media_after_bridge=true, but the re-invite appears to be done
before the confirm.js, So the media is successfully rerouted, but BEFORE the
leg b never gets hear a prompt or gets the opportunity to press 1.
To get round this I am trying to manually bypass_media in the confirm.js
script with apiExecute(uuid_media, off + session.uuid);. However only
the B leg is reinvited (and media is routed correctly). I don't see the A
leg reinvite, and then a BYE is issueed on both legs.
extension name=public_did
condition field=destination_number expression=^(12125553666)$
action application=set data=domain_name=$${domain}/
action application=set data=call_timeout=60/
action application=set data=group_confirm_key=exec/
action application=set data=group_confirm_file=javascript confirm.js/
action application=bridge data=[leg_confirm=y]
sofia/gateway/broadvox/6095553828/
/condition
/extension
/include
This is the confirm.js:
// confirm.js - FreeSwitch call confirmation script
// (c) 2009 - St‚phane Alnet
// License: GPL2 or above
console_log(info, Destination: + session.destination + \n);
if(!session.getVariable('leg_confirm'))
{
console_log(info, No need to confirm, connect the call!\n);
exit();
}
var confirmed = false;
var confirmation_digit = 1;
var try_count = 6;
var prompt_file = prompts/ToAcceptThisCallPress1.wav;
function onInput( session, type, data, arg ) {
if ( type == dtmf ) {
console_log( info, Got digit + data.digit + \n );
if ( data.digit == confirmation_digit ) {
confirmed = true;
console_log( info, Confirming session..\n );
return(false);
}
}
return(true);
}
if ( session.ready() )
{
session.answer();
session.flushDigits();
console_log(info, Starting confirmation\n);
var count = try_count;
while( session.ready() ! confirmed count-- 0 )
{
session.execute(sleep,200);
session.streamFile( prompt_file, onInput );
}
if( ! confirmed )
{
console_log(info, Not confirmed\n);
session.hangup();
}
else
{
*apiExecute(uuid_media, off + session.uuid);*
console_log(info, Confirmed\n);
}
}
else
{
console_log(info, Session is not ready.\n);
}
On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I would need to know more details about what you are doing.
you could set the variable bypass_media_after_bridge=true on the a leg
before you call the b leg and use the group_confirm feature to get the
caller
to press the key.
On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for that.
That seems to successfully re-invite and re-route the the B leg - but does
not reinvite the A leg and then immediately issues a bye on both legs.
Do I have to do something to reinvite that A leg?
On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try
apiExecute(uuid_media, off + session.uuid);
On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I was wondering whether it is possible to have FreeSwitch go into
bypass_media mode on demand?
For instance, leg a bridges to leg b - leg b is invited to accept the
call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
reroute the media) after the one is pressed.
Currently I am issuing the following from my js script that prompts for
the 1:
session.apiExecute(uuid_media,session.uuid);
Not working however.
Any help to get me going would be appreciated.
Thanks
Phillip Jones.
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--
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
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IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
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Re: [Freeswitch-users] SIP re-invite / bypass_media
Used:
session.execute(set,bypass_media_after_bridge=true);
in the confirm.js script and that works perfectly!
Thank you for you help!
On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try setting bypass_media_after_bridge=true on the session in your confirm
script
On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for responding and for your help.
The xml and confirm.js are attached below. Basically trying to
bypass_media after the leg B presses 1 to accept the call. I tried,
using bypass_media_after_bridge=true, but the re-invite appears to be done
before the confirm.js, So the media is successfully rerouted, but BEFORE the
leg b never gets hear a prompt or gets the opportunity to press 1.
To get round this I am trying to manually bypass_media in the confirm.js
script with apiExecute(uuid_media, off + session.uuid);. However only
the B leg is reinvited (and media is routed correctly). I don't see the A
leg reinvite, and then a BYE is issueed on both legs.
extension name=public_did
condition field=destination_number expression=^(12125553666)$
action application=set data=domain_name=$${domain}/
action application=set data=call_timeout=60/
action application=set data=group_confirm_key=exec/
action application=set data=group_confirm_file=javascript
confirm.js/
action application=bridge data=[leg_confirm=y]
sofia/gateway/broadvox/6095553828/
/condition
/extension
/include
This is the confirm.js:
// confirm.js - FreeSwitch call confirmation script
// (c) 2009 - St‚phane Alnet
// License: GPL2 or above
console_log(info, Destination: + session.destination + \n);
if(!session.getVariable('leg_confirm'))
{
console_log(info, No need to confirm, connect the call!\n);
exit();
}
var confirmed = false;
var confirmation_digit = 1;
var try_count = 6;
var prompt_file = prompts/ToAcceptThisCallPress1.wav;
function onInput( session, type, data, arg ) {
if ( type == dtmf ) {
console_log( info, Got digit + data.digit + \n );
if ( data.digit == confirmation_digit ) {
confirmed = true;
console_log( info, Confirming session..\n );
return(false);
}
}
return(true);
}
if ( session.ready() )
{
session.answer();
session.flushDigits();
console_log(info, Starting confirmation\n);
var count = try_count;
while( session.ready() ! confirmed count-- 0 )
{
session.execute(sleep,200);
session.streamFile( prompt_file, onInput );
}
if( ! confirmed )
{
console_log(info, Not confirmed\n);
session.hangup();
}
else
{
*apiExecute(uuid_media, off + session.uuid);*
console_log(info, Confirmed\n);
}
}
else
{
console_log(info, Session is not ready.\n);
}
On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I would need to know more details about what you are doing.
you could set the variable bypass_media_after_bridge=true on the a leg
before you call the b leg and use the group_confirm feature to get the
caller
to press the key.
On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for that.
That seems to successfully re-invite and re-route the the B leg - but
does not reinvite the A leg and then immediately issues a bye on both
legs.
Do I have to do something to reinvite that A leg?
On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try
apiExecute(uuid_media, off + session.uuid);
On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones
pjinthe...@gmail.comwrote:
Hi there,
I was wondering whether it is possible to have FreeSwitch go into
bypass_media mode on demand?
For instance, leg a bridges to leg b - leg b is invited to accept the
call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
reroute the media) after the one is pressed.
Currently I am issuing the following from my js script that prompts
for the 1:
session.apiExecute(uuid_media,session.uuid);
Not working however.
Any help to get me going would be appreciated.
Thanks
Phillip Jones.
___
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.commsn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
Re: [Freeswitch-users] SIP re-invite / bypass_media
no problem
On Thu, Jul 2, 2009 at 12:27 PM, Phillip Jones pjinthe...@gmail.com wrote:
Used:
session.execute(set,bypass_media_after_bridge=true);
in the confirm.js script and that works perfectly!
Thank you for you help!
On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try setting bypass_media_after_bridge=true on the session in your confirm
script
On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for responding and for your help.
The xml and confirm.js are attached below. Basically trying to
bypass_media after the leg B presses 1 to accept the call. I tried,
using bypass_media_after_bridge=true, but the re-invite appears to be done
before the confirm.js, So the media is successfully rerouted, but BEFORE the
leg b never gets hear a prompt or gets the opportunity to press 1.
To get round this I am trying to manually bypass_media in the confirm.js
script with apiExecute(uuid_media, off + session.uuid);. However only
the B leg is reinvited (and media is routed correctly). I don't see the A
leg reinvite, and then a BYE is issueed on both legs.
extension name=public_did
condition field=destination_number expression=^(12125553666)$
action application=set data=domain_name=$${domain}/
action application=set data=call_timeout=60/
action application=set data=group_confirm_key=exec/
action application=set data=group_confirm_file=javascript
confirm.js/
action application=bridge data=[leg_confirm=y]
sofia/gateway/broadvox/6095553828/
/condition
/extension
/include
This is the confirm.js:
// confirm.js - FreeSwitch call confirmation script
// (c) 2009 - St‚phane Alnet
// License: GPL2 or above
console_log(info, Destination: + session.destination + \n);
if(!session.getVariable('leg_confirm'))
{
console_log(info, No need to confirm, connect the call!\n);
exit();
}
var confirmed = false;
var confirmation_digit = 1;
var try_count = 6;
var prompt_file = prompts/ToAcceptThisCallPress1.wav;
function onInput( session, type, data, arg ) {
if ( type == dtmf ) {
console_log( info, Got digit + data.digit + \n );
if ( data.digit == confirmation_digit ) {
confirmed = true;
console_log( info, Confirming session..\n );
return(false);
}
}
return(true);
}
if ( session.ready() )
{
session.answer();
session.flushDigits();
console_log(info, Starting confirmation\n);
var count = try_count;
while( session.ready() ! confirmed count-- 0 )
{
session.execute(sleep,200);
session.streamFile( prompt_file, onInput );
}
if( ! confirmed )
{
console_log(info, Not confirmed\n);
session.hangup();
}
else
{
*apiExecute(uuid_media, off + session.uuid);*
console_log(info, Confirmed\n);
}
}
else
{
console_log(info, Session is not ready.\n);
}
On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I would need to know more details about what you are doing.
you could set the variable bypass_media_after_bridge=true on the a leg
before you call the b leg and use the group_confirm feature to get the
caller
to press the key.
On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for that.
That seems to successfully re-invite and re-route the the B leg - but
does not reinvite the A leg and then immediately issues a bye on both
legs.
Do I have to do something to reinvite that A leg?
On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try
apiExecute(uuid_media, off + session.uuid);
On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones pjinthe...@gmail.com
wrote:
Hi there,
I was wondering whether it is possible to have FreeSwitch go into
bypass_media mode on demand?
For instance, leg a bridges to leg b - leg b is invited to accept the
call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
reroute the media) after the one is pressed.
Currently I am issuing the following from my js script that prompts
for the 1:
session.apiExecute(uuid_media,session.uuid);
Not working however.
Any help to get me going would be appreciated.
Thanks
Phillip Jones.
___
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
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.commsn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.orgsip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
Re: [Freeswitch-users] SIP re-invite / bypass_media
try apiExecute(uuid_media, off + session.uuid); On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones pjinthe...@gmail.com wrote: Hi there, I was wondering whether it is possible to have FreeSwitch go into bypass_media mode on demand? For instance, leg a bridges to leg b - leg b is invited to accept the call by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute the media) after the one is pressed. Currently I am issuing the following from my js script that prompts for the 1: session.apiExecute(uuid_media,session.uuid); Not working however. Any help to get me going would be appreciated. Thanks Phillip Jones. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
