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[Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

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brian at freeswitch.org
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Posted: Wed Apr 15, 2009 9:16 am Post subject: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH

Can you describe the call path a bit more and what SVN rev are you on?

/b

On Apr 15, 2009, at 8:43 AM, Peter Olsson wrote:

Quote:

Brian West
brian@freeswitch.org (brian@freeswitch.org)

-- Meet us at ClueCon! http://www.cluecon.com

peter.olsson at vision...
Guest

Posted: Wed Apr 15, 2009 9:21 am Post subject: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH

When I do a call from my Avaya SIP Server to FreeSWITCH. And then let FreeSWITCH do a hangup of the call, FreeSWITCH doesn’t seem to send a ”BYE” back to the Avaya PBX. I’ve narrowed it down to this simple example in the dialplan; <action application="answer"/> <action application="sleep" data="5000"/> <action application="hangup"/> In this case no BYE is issued, and the phone still thinks the call is alive. If you want to I could send the SIP headers as well for this scenario.. Regards, Peter Olsson

anthony.minessale at g...
Guest

Posted: Wed Apr 15, 2009 10:45 am Post subject: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH

type: sofia profile internal siptrace on at the cli and try again

see if you cen see FS sending BYE to the wrong address.

This can be caused by a false positive on the NAT detection or when you need NAT mode and you don't have it enabled.

first edit the sofia profile in your config and comment out any line with the word nat in them

On Wed, Apr 15, 2009 at 8:43 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:

Quote:

When I do a call from my Avaya SIP Server to FreeSWITCH. And then let FreeSWITCH do a hangup of the call, FreeSWITCH doesn’t seem to send a ”BYE” back to the Avaya PBX. I’ve narrowed it down to this simple example in the dialplan;

      <action application="answer"/>
      <action application="sleep" data="5000"/>
      <action application="hangup"/>

In this case no BYE is issued, and the phone still thinks the call is alive. If you want to I could send the SIP headers as well for this scenario..

Regards,

Peter Olsson

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
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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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
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FreeSWITCH Developer Conference
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anthony.minessale at g...
Guest

Posted: Wed Apr 15, 2009 12:00 pm Post subject: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH

This sounds familiar:

What revision of the code is this?
Can you confirm you have this problem with SVN trunk (r13034 at the time of this writing).

On Wed, Apr 15, 2009 at 11:24 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:

Quote:

This is the full SIP-trace for the call. It’s not sending a BYE at all, and I can’t see one in Wireshark either. As you can see in the end there is a call to hangup_function(), but no SIP messages after that. When I manually hangup the phone I can see it sends BYE to FreeSWITCH (which is quite expected, since it thinks the call still exists), and FreeSWITCH just answers ”481 Call Does Not Exist” – which of course is also expected, since the call was dropped.

recv 1255 bytes from udp/[192.168.94.53]:32769 at 16:17:57.853727:
   ------------------------------------------------------------------------
   INVITE sip:2100@192.168.1.155:5060;lr SIP/2.0
   Accept-Language: en
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 INVITE
   From: "Peter Olsson" <sip:1002@sip.se:6001>;tag=80948a675733de13449f79df00
   Record-Route: <sip:192.168.94.53:5060;lr>,<sip:192.168.94.53:6001;lr;transport=tls>
   To: "2100" <sip:2100@192.168.94.53 ([email]sip%3A2100@192.168.94.53[/email])>
   Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00
   Content-Length: 165
   Content-Type: application/sdp
   Contact: "Peter Olsson" <sip:1002@192.168.94.53:6001;transport=tls>
   Max-Forwards: 67
   User-Agent: Avaya CM/R015x.01.1.415.1
   Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
   Supported: 100rel,timer,replaces,join,histinfo
   Alert-Info: <cid:internal@invalid.unknown.domain>;avaya-cm-alert-type=internal
   Min-SE: 1200
   Session-Expires: 1200;refresher=uac
   P-Asserted-Identity: "Peter Olsson" <sip:1002@sip.se:6001>
   History-Info: <sip:2100@192.168.94.53 ([email]sip%3A2100@192.168.94.53[/email])>;index=1,"2100" <sip:2100@192.168.94.53 ([email]sip%3A2100@192.168.94.53[/email])>;index=1.1

   v=0
   o=- 1 1 IN IP4 192.168.94.53
   s=-
   c=IN IP4 192.168.94.59
   b=AS:64
   t=0 0
   m=audio 2062 RTP/AVP 8 127
   a=rtpmap:8 PCMA/8000
   a=rtpmap:127 telephone-event/8000
   ------------------------------------------------------------------------
send 541 bytes to udp/[192.168.94.53]:5060 at 16:17:57.854727:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00
   Record-Route: <sip:192.168.94.53:5060;lr>
   Record-Route: <sip:192.168.94.53:6001;lr;transport=tls>
   From: "Peter Olsson" <sip:1002@sip.se:6001>;tag=80948a675733de13449f79df00
   To: "2100" <sip:2100@192.168.94.53 ([email]sip%3A2100@192.168.94.53[/email])>
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Content-Length: 0

   ------------------------------------------------------------------------
2009-04-15 18:17:57 [NOTICE] switch_channel.c:597 switch_channel_set_name() NewChannel sofia/internal/1002@sip.se:6001 [fa1c328e-bdfe-7d49-ab6f-dc9ec791c455]
2009-04-15 18:17:57 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Peter Olsson->2100 in context public
2009-04-15 18:17:57 [NOTICE] mod_dptools.c:649 answer_function() Channel [sofia/internal/1002@sip.se:6001] has been answered
send 1322 bytes to udp/[192.168.94.53]:5060 at 16:17:57.871727:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00
   Record-Route: <sip:192.168.94.53:5060;lr>
   Record-Route: <sip:192.168.94.53:6001;lr;transport=tls>
   From: "Peter Olsson" <sip:1002@sip.se:6001>;tag=80948a675733de13449f79df00
   To: "2100" <sip:2100@192.168.94.53 ([email]sip%3A2100@192.168.94.53[/email])>;tag=Sv6KrDv9vQrer
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 INVITE
   Contact: <sip:mod_sofia@192.168.1.155:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Require: timer
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Session-Expires: 1200;refresher=uac
   Min-SE: 1200
   Content-Type: application/sdp
Content-Disposition: session
   Content-Length: 265

   v=0
   o=FreeSWITCH 484797194364394181 220756314446402535 IN IP4 192.168.1.155
   s=FreeSWITCH
   c=IN IP4 192.168.1.155
   t=0 0
   m=audio 23574 RTP/AVP 8 127
   a=rtpmap:8 PCMA/8000
   a=rtpmap:127 telephone-event/8000
   a=fmtp:127 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   ------------------------------------------------------------------------
recv 521 bytes from udp/[192.168.94.53]:32769 at 16:17:57.880727:
   ------------------------------------------------------------------------
   ACK sip:mod_sofia@192.168.1.155:5060;transport=udp SIP/2.0
   From: "Peter Olsson" <sip:1002@sip.se:6001>;tag=80948a675733de13449f79df00
   To: "2100" <sip:2100@192.168.94.53 ([email]sip%3A2100@192.168.94.53[/email])>;tag=Sv6KrDv9vQrer
   Call-ID: 80948a675733de14449f79df00
   CSeq: 1 ACK
   Max-Forwards: 69
   Via: SIP/2.0/UDP 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.1,SIP/2.0/TLS 192.168.94.53:6001;psrrposn=1;branch=z9hG4bK80948a675733de16449f79df00

   User-Agent: Avaya CM/R015x.01.1.415.1
   Content-Length: 0
   Record-Route: <sip:192.168.94.53:5060;lr>

   ------------------------------------------------------------------------
2009-04-15 18:18:02 [NOTICE] mod_dptools.c:633 hangup_function() Hangup sofia/internal/1002@sip.se:6001 [CS_EXECUTE] [NORMAL_CLEARING]
2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1021 switch_core_session_thread() Session 5 (sofia/internal/1002@sip.se:6001) Ended
2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1023 switch_core_session_thread() Close Channel sofia/internal/1002@sip.se:6001 [CS_DESTROY]

Från: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] För Anthony Minessale
Skickat: den 15 april 2009 17:27
Till: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?

type: sofia profile internal siptrace on at the cli and try again

see if you cen see FS sending BYE to the wrong address.

This can be caused by a false positive on the NAT detection or when you need NAT mode and you don't have it enabled.

first edit the sofia profile in your config and comment out any line with the word nat in them

On Wed, Apr 15, 2009 at 8:43 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
When I do a call from my Avaya SIP Server to FreeSWITCH. And then let FreeSWITCH do a hangup of the call, FreeSWITCH doesn’t seem to send a ”BYE” back to the Avaya PBX. I’ve narrowed it down to this simple example in the dialplan;

      <action application="answer"/>
      <action application="sleep" data="5000"/>
      <action application="hangup"/>

In this case no BYE is issued, and the phone still thinks the call is alive. If you want to I could send the SIP headers as well for this scenario..

Regards,

Peter Olsson

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

!DSPAM:49e5fe5232932637379622!

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peter.olsson at vision...
Guest

Posted: Thu Apr 16, 2009 2:26 am Post subject: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH

Allright, I tried this again now, with revision 13042 – it’s the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I’ll try to help out as much as possible. Peter Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West Skickat: den 15 april 2009 23:21 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West brian@freeswitch.org (brian@freeswitch.org) -- Meet us at ClueCon! http://www.cluecon.com !DSPAM:49e651b332933023977319!

anthony.minessale at g...
Guest

Posted: Thu Apr 16, 2009 7:36 am Post subject: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH

yes open a jira http://jira.freeswitch.org

*attach* the following (do not paste it inline into the comments and give all trace files a .txt extension)

repeat the trace you did earlier with more debugging enabled.
     type these 3 cli commands before you call
     sofia profile internal siptrace on
     sofia loglevel all 9
     console loglevel debug

On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:

Quote:

Allright, I tried this again now, with revision 13042 – it’s the same result as before.. Should I file a jira case for this?

If you want any more information, or more traces, please get back to me, and I’ll try to help out as much as possible.

Peter

Från: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] För Brian West
Skickat: den 15 april 2009 23:21
Till: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)

Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call?

What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both.

/b

On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:

I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that.

Just to make the scenario a bit more clear;
The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE.

Peter

Brian West

brian@freeswitch.org (brian@freeswitch.org)

-- Meet us at ClueCon! http://www.cluecon.com

!DSPAM:49e651b332933023977319!

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

peter.olsson at vision...
Guest

Posted: Thu Apr 16, 2009 10:04 am Post subject: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH

I’ve added this as jira case http://jira.freeswitch.org/browse/MODSOFIA-4 I wasn’t sure if it should be under mod_sofia or sofia-sip. The report has a full debug log attached. Regards, Peter Olsson Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 16 april 2009 14:23 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? yes open a jira http://jira.freeswitch.org attach the following (do not paste it inline into the comments and give all trace files a .txt extension) repeat the trace you did earlier with more debugging enabled. type these 3 cli commands before you call sofia profile internal siptrace on sofia loglevel all 9 console loglevel debug On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote: Allright, I tried this again now, with revision 13042 – it’s the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I’ll try to help out as much as possible. Peter Från: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] För Brian West Skickat: den 15 april 2009 23:21 Till: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org) Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West brian@freeswitch.org (brian@freeswitch.org) -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (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_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email]) GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email]) IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email]) iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email]) pstn:213-799-1400 !DSPAM:49e725b432939831339029!

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