When using nimbuzz, call disconnects after 90 seconds with RECOVERY_ON_TIMER_EXPIRE
Please help, While other SIP clients works well, there is a problem with nimbuzz SIP client, which quite frequently used by 'not-advanced' users. seems It which works through their services (asterisk 1.6) This problem was not happening while I have been using Asterisk too, but after recent migration to FS - it does. nimbuzz client stays like online, just sound disappears, but there is a confirmation from nimbuzz server side, regarding the BYE most likely a NAT issue, but if the same time using Asterisk on the same server - there is not problems. So might be there any options to tune? there are some SIP activity like this before calls disconnects, no replies: send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: ------------------------------------------------------------------------ INVITE sip:anton@192.168.0.20:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe Max-Forwards: 70 From: "3777077" <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS To: "anton@62.122.137.32" <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 26214721 INVITE Contact: <sip:3777077@62.122.137.32:5060;transport=udp> User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 272 v=0 o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 s=FreeSWITCH c=IN IP4 62.122.137.32 t=0 0 m=audio 20062 RTP/AVP 18 106 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:20 and than 2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup sofia/internal/anton@62.122.137.32 [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE] recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0 ------------------------------------------------------------------------
In a case like this I like to compare working siptraces against non-working ones so that I have a frame of reference. Put those on pastebin and the gang here will take a look. -MC On Fri, Mar 30, 2012 at 3:16 AM, Anton VG <anton.vazir@gmail.com> wrote:
Please help,
While other SIP clients works well, there is a problem with nimbuzz SIP client, which quite frequently used by 'not-advanced' users. seems It which works through their services (asterisk 1.6)
This problem was not happening while I have been using Asterisk too, but after recent migration to FS - it does.
nimbuzz client stays like online, just sound disappears, but there is a confirmation from nimbuzz server side, regarding the BYE
most likely a NAT issue, but if the same time using Asterisk on the same server - there is not problems. So might be there any options to tune?
there are some SIP activity like this before calls disconnects, no replies:
send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: ------------------------------------------------------------------------ INVITE sip:anton@192.168.0.20:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe Max-Forwards: 70 From: "3777077" <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS To: "anton@62.122.137.32" <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 26214721 INVITE Contact: <sip:3777077@62.122.137.32:5060;transport=udp> User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 272
v=0 o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 s=FreeSWITCH c=IN IP4 62.122.137.32 t=0 0 m=audio 20062 RTP/AVP 18 106 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:20
and than
2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup sofia/internal/anton@62.122.137.32 [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE]
recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061 ;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0
------------------------------------------------------------------------ recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061 ;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0
------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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As a start try disabling session timers in your FreeSWITCH profile: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer On Fri, Mar 30, 2012 at 6:16 AM, Anton VG <anton.vazir@gmail.com> wrote:
Please help,
While other SIP clients works well, there is a problem with nimbuzz SIP client, which quite frequently used by 'not-advanced' users. seems It which works through their services (asterisk 1.6)
This problem was not happening while I have been using Asterisk too, but after recent migration to FS - it does.
nimbuzz client stays like online, just sound disappears, but there is a confirmation from nimbuzz server side, regarding the BYE
most likely a NAT issue, but if the same time using Asterisk on the same server - there is not problems. So might be there any options to tune?
there are some SIP activity like this before calls disconnects, no replies:
send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: ------------------------------------------------------------------------ INVITE sip:anton@192.168.0.20:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe Max-Forwards: 70 From: "3777077" <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS To: "anton@62.122.137.32" <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 26214721 INVITE Contact: <sip:3777077@62.122.137.32:5060;transport=udp> User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 272
v=0 o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 s=FreeSWITCH c=IN IP4 62.122.137.32 t=0 0 m=audio 20062 RTP/AVP 18 106 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:20
and than
2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup sofia/internal/anton@62.122.137.32 [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE]
recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0
------------------------------------------------------------------------ recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0
------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
Thanks for help guys! Disabling SIP timers solves this problem. But I suppose it can raise other problems in other cases. Michael, do you think collecting of SIP headers still reasonable, or Asterisk just ignores or do not have any specific SIP session timer? Can I enable/disable timers per session rather per profile (say via the channel variable while 'originating'? in this case I can check if nimbuss is a UA and than disable timers? 2012/3/31 Kristian Kielhofner <kris@kriskinc.com>:
As a start try disabling session timers in your FreeSWITCH profile:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
On Fri, Mar 30, 2012 at 6:16 AM, Anton VG <anton.vazir@gmail.com> wrote:
Please help,
While other SIP clients works well, there is a problem with nimbuzz SIP client, which quite frequently used by 'not-advanced' users. seems It which works through their services (asterisk 1.6)
This problem was not happening while I have been using Asterisk too, but after recent migration to FS - it does.
nimbuzz client stays like online, just sound disappears, but there is a confirmation from nimbuzz server side, regarding the BYE
most likely a NAT issue, but if the same time using Asterisk on the same server - there is not problems. So might be there any options to tune?
there are some SIP activity like this before calls disconnects, no replies:
send 977 bytes to udp/[195.211.48.6]:12621 at 10:00:02.535856: ------------------------------------------------------------------------ INVITE sip:anton@192.168.0.20:5061;transport=udp SIP/2.0 Via: SIP/2.0/UDP 62.122.137.32;rport;branch=z9hG4bKaQm6tK1aeZ7Fe Max-Forwards: 70 From: "3777077" <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS To: "anton@62.122.137.32" <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 26214721 INVITE Contact: <sip:3777077@62.122.137.32:5060;transport=udp> User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 272
v=0 o=FreeSWITCH 1333081454 1333081455 IN IP4 62.122.137.32 s=FreeSWITCH c=IN IP4 62.122.137.32 t=0 0 m=audio 20062 RTP/AVP 18 106 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:106 telephone-event/8000 a=fmtp:106 0-16 a=silenceSupp:off - - - - a=ptime:20
and than
2012-03-30 15:00:03.554460 [NOTICE] sofia.c:6301 Hangup sofia/internal/anton@62.122.137.32 [CS_SOFT_EXECUTE] [RECOVERY_ON_TIMER_EXPIRE]
recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960518: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960672: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0
------------------------------------------------------------------------ recv 331 bytes from udp/[195.211.48.6]:12621 at 10:00:13.960734: ------------------------------------------------------------------------ BYE sip:3777077@62.122.137.32:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ send 480 bytes to udp/[195.211.48.6]:12621 at 10:00:13.960781: ------------------------------------------------------------------------ SIP/2.0 481 Call Does Not Exist Via: SIP/2.0/UDP 192.168.0.20:5061;rport=12621;branch=z9hG4bK5207427786;received=195.211.48.6 From: <sip:anton@62.122.137.32>;tag=sip1331131792771 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS Call-ID: 3696882418 CSeq: 3 BYE User-Agent: Synaptic Switch v2.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0
------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079094: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------ recv 365 bytes from udp/[195.211.48.6]:12621 at 10:00:14.079293: ------------------------------------------------------------------------ ACK sip:anton@62.122.137.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5061;rport;branch=z9hG4bK5207427786 Max-Forwards: 70 To: <sip:3777077@62.122.137.32>;tag=eKv1Nej7ZrNUS From: <sip:anton@62.122.137.32>;tag=sip1331131792771 Call-ID: 3696882418 CSeq: 3 ACK Contact: <sip:anton@192.168.0.20:5061> Expires: 3600 User-Agent: Nimbuzz Single Content-Length: 0
------------------------------------------------------------------------
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kristian Kielhofner
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
participants (3)
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Anton VG
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Kristian Kielhofner
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Michael Collins