<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">FreeSWITCH debug level logs should help tell you exactly what is killing the call.<div><br></div><div><br><div><div>On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:</div><br class="Apple-interchange-newline"><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">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">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/">209.216.2.211:5060</a></div>
<div>tport_resolve addrinfo = <a href="http://209.216.2.211:5060/">209.216.2.211:5060</a></div><div>tport_by_addrinfo(0x18413c0): not found by name */<a href="http://209.216.2.211:5060/">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/">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 href="sip:199.173.100.16:5060;transport=UDP">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 href="sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on">sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on</a>></div><div> Route: <<a href="sip:65.211.120.237:5060;lr">sip:65.211.120.237:5060;lr</a>></div>
<div> Route: <<a href="sip:63.110.102.239;lr">sip:63.110.102.239;lr</a>></div><div> Max-Forwards: 70</div><div> From: <<a href="sip:+12133304391@63.110.102.239:5060;user=phone">sip:+12133304391@63.110.102.239:5060;user=phone</a>>;tag=cgBe054jZrt3a</div><div> To: <<a href="sip:+14158867717@199.173.100.16:5060;user=phone">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 href="sip:+12133304391@67.220.216.146:5080;transport=udp">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">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; position: static; z-index: auto; ">
<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 class="h5"><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 href="sip:199.173.100.144:5060;transport=UDP">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 href="sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on">sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on</a>></div><div>Route: <<a href="sip:65.217.40.210:5060;lr">sip:65.217.40.210:5060;lr</a>></div><div>Route: <<a href="sip:63.110.102.238;lr">sip:63.110.102.238;lr</a>></div><div>Max-Forwards: 70</div><div>From: <<a href="sip:+1212381XXXX@63.110.102.238:5060;user=phone">sip:+1212381XXXX@63.110.102.238:5060;user=phone</a>>;tag=Ztr5ycrv3QZ1g</div>
<div>To: <<a href="sip:+1909635XXXX@199.173.100.144:5060;user=phone">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 href="sip:+1212381XXXX@66.197.142.69:5080;transport=udp">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 href="sip:199.173.100.144:5060;transport=UDP">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 href="sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on">sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on</a>></div>
<div>Route: <<a href="sip:65.217.40.210:5060;lr">sip:65.217.40.210:5060;lr</a>></div><div>Route: <<a href="sip:63.110.102.238;lr">sip:63.110.102.238;lr</a>></div><div>Max-Forwards: 70</div><div>From: <<a href="sip:+1212381XXXX@63.110.102.238:5060;user=phone">sip:+1212381XXXX@63.110.102.238:5060;user=phone</a>>;tag=Ztr5ycrv3QZ1g</div><div>
To: <<a href="sip:+1909635XXXX@199.173.100.144:5060;user=phone">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 href="sip:+1212381XXXX@66.197.142.69:5080;transport=udp">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; position: static; z-index: auto; "> 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></div></div></blockquote></div></div></blockquote></div></div></body></html>