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<P><FONT SIZE=2>I am dealing with the same problem... Brian, there is something wrong here. The effect is my calls get terminated every two minutes +hen my sipura re-registers. I have a trace with full debug captured if you want to see.<BR>
<BR>
Michael Murdock<BR>
VP Operations & Development<BR>
Switchmaxx Product Line<BR>
125 N. Market. Suite 1520<BR>
Wichita KS. 67202<BR>
(316) 932-2693<BR>
<BR>
-----Original Message-----<BR>
From: Brian West [<A HREF="mailto:brian.west@mac.com">mailto:brian.west@mac.com</A>]<BR>
Sent: Tuesday, May 29, 2007 10:14 AM Eastern Standard Time<BR>
To: freeswitch-dev@lists.freeswitch.org<BR>
Subject: Re: [Freeswitch-dev] SIP Timeout waiting for 200 ACK<BR>
<BR>
I think that since we are a B2BUA tech_pvt->profile->url is the right <BR>
way. The other way is correct if you're a proxy. Remember <BR>
FreeSWITCH NOT a proxy.<BR>
<BR>
/b<BR>
<BR>
On May 29, 2007, at 8:58 AM, freeswitch@dalethatcher.com wrote:<BR>
<BR>
> I'm seeing an issue during a bridge where the '200' from freeswitch is<BR>
> not being ACK'd and Freeswitch is correctly dropping the connection.<BR>
> However I belive that the Contact header in the 200 packet is causing<BR>
> the problem. Am I missing a config parameter?<BR>
><BR>
> If I back out the revision 5208 change to mod_sofia.c from the trunk<BR>
> source then the problem goes away. (SIPTAG_CONTACT_STR(tech_pvt-<BR>
> >to_uri)<BR>
> back to SIPTAG_CONTACT_STR(tech_pvt->profile->url)).<BR>
><BR>
> My test setup is (Freeswitch is behind a NAT):<BR>
><BR>
> External SIP -> Freeswitch -> Gizmo<BR>
><BR>
> The freeswitch config for the extension is:<BR>
><BR>
> <extension name="test"><BR>
> <condition field="destination_number" expression="^test$"><BR>
> <action application="bridge" data="sofia/gizmo/<BR>
> 17470355893@proxy01.sipphone.com"/><BR>
> </condition><BR>
> </extension><BR>
><BR>
> I've put a full copy of the log here:<BR>
><BR>
> <A HREF="http://pastebin.freeswitch.org/3008">http://pastebin.freeswitch.org/3008</A><BR>
><BR>
> My internet facing profile is:<BR>
><BR>
> <profile name="internet"><BR>
> <settings><BR>
> <param name="debug" value="1"/><BR>
> <param name="rfc2833-pt" value="101"/><BR>
> <param name="sip-port" value="5060"/><BR>
> <param name="dialplan" value="XML"/><BR>
> <param name="dtmf-duration" value="100"/><BR>
> <param name="codec-prefs" value="$${global_codec_prefs}"/><BR>
> <param name="codec-ms" value="20"/><BR>
> <param name="use-rtp-timer" value="true"/><BR>
> <param name="rtp-timer-name" value="soft"/><BR>
> <param name="rtp-ip" value="$${bind_server_ip}"/><BR>
> <param name="sip-ip" value="$${bind_server_ip}"/><BR>
> <param name="accept-blind-reg" value="true"/><BR>
> <param name="nonce-ttl" value="60"/><BR>
> <param name="ext-rtp-ip" value="80.189.98.58"/><BR>
> <param name="ext-sip-ip" value="80.189.98.58"/><BR>
> <param name="sip-domain" value="sip.dalethatcher.com"/><BR>
> </settings><BR>
> </profile><BR>
><BR>
> With the current trunk the SIP packet that does not get a response is:<BR>
><BR>
> send 1182 bytes to udp/[193.111.201.32]:5060 at 13:05:15.136189:<BR>
> <BR>
> ----------------------------------------------------------------------<BR>
> --<BR>
> SIP/2.0 200 OK<BR>
> Via: SIP/2.0/UDP 193.111.201.32;branch=z9hG4bK614f.db2cc426.0<BR>
> Via: SIP/2.0/UDP 193.111.200.182;branch=z9hG4bK614f.9dbef354.0<BR>
> Via: SIP/2.0/UDP <BR>
> 194.145.189.10:3670;branch=z9hG4bK00E0F556495005A857E00000230F<BR>
> Record-Route: <sip:<BR>
> 193.111.201.32;lr=on;ftag=00E0F556495005A857E00000222A><BR>
> Record-Route: <sip:<BR>
> 193.111.200.182;lr=on;ftag=00E0F556495005A857E00000222A><BR>
> From: <sip:<BR>
> 02085555129@194.145.189.10>;tag=00E0F556495005A857E00000222A<BR>
> To: <sip:442070968842@193.111.200.182>;tag=r5ae93vm8aK1p<BR>
> Call-ID: 00E0F556495005A857E000001549@194.145.189.10<BR>
> CSeq: 35816 INVITE<BR>
> Contact: <sip:442070968842@193.111.200.182:5060><BR>
> User-Agent: FreeSWITCH(mod_sofia)<BR>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, <BR>
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO<BR>
> Supported: 100rel, precondition<BR>
> Content-Type: application/sdp<BR>
> Content-Disposition: session<BR>
> Content-Length: 290<BR>
><BR>
> v=0<BR>
> o=FreeSWITCH 6489556414011969222 9057524047013517960 IN IP4 <BR>
> 192.168.93.5<BR>
> s=FreeSWITCH<BR>
> c=IN IP4 80.189.98.58<BR>
> t=0 0<BR>
> a=sendrecv<BR>
> m=audio 10002 RTP/AVP 0 96 13<BR>
> a=rtpmap:0 PCMU/8000<BR>
> a=rtpmap:96 telephone-event/8000<BR>
> a=fmtp:96 0-16<BR>
> a=rtpmap:13 CN/8000<BR>
> a=ptime:20<BR>
> m=image 0 UDPTL 9<BR>
> <BR>
> ----------------------------------------------------------------------<BR>
> --<BR>
><BR>
> Backing out the change modifies the Contact header to:<BR>
><BR>
> Contact: <sip:mod_sofia@80.189.98.58:5060><BR>
><BR>
> Which gets an ACK packet and doesn't drop the call.<BR>
><BR>
> thanks,<BR>
><BR>
> - Dale<BR>
><BR>
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