<div><div>The debug level logs to the console, right? The pastebin, had log level debug, sofia trace on for external and default, and sofia loglevel all 9. Is there another log enable command I'm missing? It seems loglevel all 9 outputs enter and exit functions, but at least to my novice eye, it's not too obvious why freeswitch is sending a BYE to my DID provider.</div>
<div><br></div><div>I did do some additional testing, and my prior comment about it working in the internal profile is incorrect. Even if I put my DID provider in my internal profile, I still receive the Exchange_Routing_Error after being bridged with a few channels. However, I purchased a DID from icall, and that DID worked. So icall YES, <a href="http://didforsale.com">didforsale.com</a> NO...It's too bad tho that <a href="http://didforsale.com">didforsale.com</a> doesn't work too well with FreeSWITCH because their 20 unlimited inbound channels can't be beat. </div>
<div><br></div><div>--matt</div><div><a href="http://www.hellohunter.com">http://www.hellohunter.com</a></div><div><br><br><div class="gmail_quote">On Mon, Oct 19, 2009 at 8:56 AM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com">mike@jerris.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div style="word-wrap:break-word">FreeSWITCH debug level logs should help tell you exactly what is killing the call.<div>
<div></div><div class="h5"><div><br></div><div><br><div><div>On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:</div><br><blockquote type="cite">I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects for users utilizing my inbound DID to connect to my FreeSWITCH server. It's a predictive dialing application, with one agent session being bridged with multiple calls and transfered back and forth between extensions in my dial plan. After a random number of bridging and transferring, FreeSWITCH suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It does not happen at any one-point in my dial plan, or applications--it just randomly disconnects when a call that the Agent is bridged to hangs-up or is disconnected. It seems to only happen when two external sip profiles are being bridged together, and not when an internal and external profile is being bridged.<div>
<br></div><div>I turned</div><div><br></div><div>sip trace on and</div><div>sofia loglevel all 9</div><div><br></div><div>below is the the snippet. I've posted the entire Agent session at the following pastebin <a href="http://pastebin.freeswitch.org/10756" target="_blank">http://pastebin.freeswitch.org/10756</a> </div>
<div><br></div><div><div>tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/<a href="http://208.76.18.254:5080/sip" target="_blank">208.76.18.254:5080/sip</a> next=(nil)</div><div>nta: received 200 OK for BYE (121818983)</div>
<div>nta: 200 OK is going to a transaction</div><div>nta_outgoing: RTT is 84.409 ms</div><div>tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30</div><div>nua(0x1ad6fb0): event r_bye 200 OK</div><div>nua(0x1ad6fb0): call state changed: terminating -> terminated</div>
<div>nua(0x1ad6fb0): event i_state 200 to BYE</div><div>nua: nua_application_event: entering</div><div>nua(0x1ad6fb0): event i_terminated 200 to BYE</div><div>nua: nua_handle_magic: entering</div><div>nua(0x1ad6fb0): removing session usage</div>
<div>soa_destroy(static::0x1b5ae90) called</div><div>nua: nua_application_event: entering</div><div>nta_leg_destroy(0x1b594a0)</div><div>nua: nua_handle_magic: entering</div><div>nua: nua_handle_bind: entering</div><div>
nua: nua_application_event: entering</div>
<div>nua: nua_handle_magic: entering</div><div>nua: terminated session 0x1ad6fb0</div><div>nua: nua_handle_destroy: entering</div><div>nua(0x1ad6fb0): recv signal r_destroy</div><div>nta_leg_destroy((nil))</div><div>nua(0x1ad6fb0): sent signal r_destroy</div>
<div>nta: timer set next to 28 ms</div><div>nta: timer E fired, retransmit BYE (121818989)</div><div>tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil)</div><div>tport_tsend(0x18413c0) tpn = */<a href="http://209.216.2.211:5060/" target="_blank">209.216.2.211:5060</a></div>
<div>tport_resolve addrinfo = <a href="http://209.216.2.211:5060/" target="_blank">209.216.2.211:5060</a></div><div>tport_by_addrinfo(0x18413c0): not found by name */<a href="http://209.216.2.211:5060/" target="_blank">209.216.2.211:5060</a></div>
<div>tport_vsend(0x18413c0): 862 bytes of 862 to udp/<a href="http://209.216.2.211:5060/" target="_blank">209.216.2.211:5060</a></div>
<div>tport_vsend returned 862</div><div>send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690:</div><div> ------------------------------------------------------------------------</div><div> BYE <a>sip:199.173.100.16:5060;transport=UDP</a> SIP/2.0</div>
<div> Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN</div><div> Route: <<a>sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on</a>></div><div> Route: <<a>sip:65.211.120.237:5060;lr</a>></div>
<div> Route: <<a>sip:63.110.102.239;lr</a>></div><div> Max-Forwards: 70</div><div> From: <<a>sip:+12133304391@63.110.102.239:5060;user=phone</a>>;tag=cgBe054jZrt3a</div><div> To: <<a>sip:+14158867717@199.173.100.16:5060;user=phone</a>>;tag=4adc7-13c4-1ab03-71ce3705-1ab03</div>
<div> Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124</div><div> CSeq: 121818989 BYE</div><div> Contact: <<a>sip:+12133304391@67.220.216.146:5080;transport=udp</a>></div><div> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135</div>
<div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY</div><div> Supported: timer, precondition, path, replaces</div><div> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"</div>
<div> Content-Length: 0</div><div><br></div><div> ------------------------------------------------------------------------</div><div><br></div><div>Thanks.</div><div>--matt</div></div><div><br><div class="gmail_quote">
On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color:rgb(204, 204, 204);border-left-style:solid;padding-left:1ex">
<div style="word-wrap:break-word"><a href="http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP" target="_blank">http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP</a><div><br></div><div>turn the logging all the way up and see what it says.</div>
<div><br></div><div>Mike</div><div><div></div><div><div><br><div><div>On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:</div><br><blockquote type="cite"><span style="border-collapse:collapse">Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the logs below, but I am still at a loss at being able to identify the error or reproduce it consistently. The below log indicates to me that my FS server is initiating sending 2 BYE message to my DID provider (<a href="http://didforsale.com/" style="color:rgb(20, 125, 186)" target="_blank">didforsale.com</a>). Is there a way I can look further inside FreeSWITCH to see why it is sending this BYE packet?<div>
<br></div><div><br></div><div><div>sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:</div><div>BYE <a>sip:199.173.100.144:5060;transport=UDP</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD</div>
<div>Route: <<a>sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on</a>></div><div>Route: <<a>sip:65.217.40.210:5060;lr</a>></div><div>Route: <<a>sip:63.110.102.238;lr</a>></div><div>Max-Forwards: 70</div>
<div>From: <<a>sip:+1212381XXXX@63.110.102.238:5060;user=phone</a>>;tag=Ztr5ycrv3QZ1g</div>
<div>To: <<a>sip:+1909635XXXX@199.173.100.144:5060;user=phone</a>>;tag=dc7-13c4-2401b7-46dea593-2401b7</div><div>Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441</div><div>CSeq: 118584736 BYE</div>
<div>
Contact: <<a>sip:+1212381XXXX@66.197.142.69:5080;transport=udp</a>></div><div>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO</div>
<div>Supported: timer, precondition, path, replaces</div><div>Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"</div><div>Content-Length: 0</div><div><br></div></div><div><br></div><div><div>sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:</div>
<div>BYE <a>sip:199.173.100.144:5060;transport=UDP</a> SIP/2.0</div><div>Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD</div><div>Route: <<a>sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on</a>></div>
<div>Route: <<a>sip:65.217.40.210:5060;lr</a>></div><div>Route: <<a>sip:63.110.102.238;lr</a>></div><div>Max-Forwards: 70</div><div>From: <<a>sip:+1212381XXXX@63.110.102.238:5060;user=phone</a>>;tag=Ztr5ycrv3QZ1g</div>
<div>
To: <<a>sip:+1909635XXXX@199.173.100.144:5060;user=phone</a>>;tag=dc7-13c4-2401b7-46dea593-2401b7</div> <div>Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441</div><div>CSeq: 118584736 BYE</div><div>
Contact: <<a>sip:+1212381XXXX@66.197.142.69:5080;transport=udp</a>></div>
<div>User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M</div> <div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO</div><div>Supported: timer, precondition, path, replaces</div>
<div>Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"</div> <div>Content-Length: 0</div></div></span><br><div class="gmail_quote">On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <span dir="ltr"><<a href="mailto:mrene_lists@avgs.ca" target="_blank">mrene_lists@avgs.ca</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color:rgb(204, 204, 204);border-left-style:solid;padding-left:1ex"> Hi,<br> <br>
Digging a bit in mod_sofia releaved that it can be caused by a SIP<br> code 482 (loop detected), 483 (too many hops) or 484 (address<br> incomplete).<br> <br> Do a SIP trace to sched more light on what's happening.<br>
<br> Mathieu Rene<br> Avant-Garde Solutions Inc<br> Office: + 1 (514) 664-1044 x100<br> Cell: +1 (514) 664-1044 x200<br> <a href="mailto:mrene@avgs.ca" target="_blank">m</a></blockquote></div></blockquote></div></div></div>
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