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