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. 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, didforsale.com NO...It's too bad tho that didforsale.com doesn't work too well with FreeSWITCH because their 20 unlimited inbound channels can't be beat. --matt http://www.hellohunter.com On Mon, Oct 19, 2009 at 8:56 AM, Michael Jerris <mike@jerris.com> wrote:
FreeSWITCH debug level logs should help tell you exactly what is killing the call.
On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:
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. I turned
sip trace on and sofia loglevel all 9
below is the the snippet. I've posted the entire Agent session at the following pastebin http://pastebin.freeswitch.org/10756
tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/ 208.76.18.254:5080/sip next=(nil) nta: received 200 OK for BYE (121818983) nta: 200 OK is going to a transaction nta_outgoing: RTT is 84.409 ms tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30 nua(0x1ad6fb0): event r_bye 200 OK nua(0x1ad6fb0): call state changed: terminating -> terminated nua(0x1ad6fb0): event i_state 200 to BYE nua: nua_application_event: entering nua(0x1ad6fb0): event i_terminated 200 to BYE nua: nua_handle_magic: entering nua(0x1ad6fb0): removing session usage soa_destroy(static::0x1b5ae90) called nua: nua_application_event: entering nta_leg_destroy(0x1b594a0) nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: terminated session 0x1ad6fb0 nua: nua_handle_destroy: entering nua(0x1ad6fb0): recv signal r_destroy nta_leg_destroy((nil)) nua(0x1ad6fb0): sent signal r_destroy nta: timer set next to 28 ms nta: timer E fired, retransmit BYE (121818989) tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil) tport_tsend(0x18413c0) tpn = */209.216.2.211:5060 tport_resolve addrinfo = 209.216.2.211:5060 tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060 tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060 tport_vsend returned 862 send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690: ------------------------------------------------------------------------ BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN Route: <sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on> Route: <sip:65.211.120.237:5060;lr> Route: <sip:63.110.102.239;lr> Max-Forwards: 70 From: <sip:+12133304391@63.110.102.239:5060;user=phone
;tag=cgBe054jZrt3a To: <sip:+14158867717@199.173.100.16:5060;user=phone ;tag=4adc7-13c4-1ab03-71ce3705-1ab03 Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124 CSeq: 121818989 BYE Contact: <sip:+12133304391@67.220.216.146:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR" Content-Length: 0
------------------------------------------------------------------------
Thanks. --matt
On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris <mike@jerris.com> wrote:
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP turn the logging all the way up and see what it says.
Mike
On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
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 (didforsale.com). Is there a way I can look further inside FreeSWITCH to see why it is sending this BYE packet?
sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208: BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on> Route: <sip:65.217.40.210:5060;lr> Route: <sip:63.110.102.238;lr> Max-Forwards: 70 From: <sip:+1212381XXXX@63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g To: <sip:+1909635XXXX@199.173.100.144:5060;user=phone
;tag=dc7-13c4-2401b7-46dea593-2401b7 Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441 CSeq: 118584736 BYE Contact: <sip:+1212381XXXX@66.197.142.69:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR" Content-Length: 0
sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589: BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on> Route: <sip:65.217.40.210:5060;lr> Route: <sip:63.110.102.238;lr> Max-Forwards: 70 From: <sip:+1212381XXXX@63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g To: <sip:+1909635XXXX@199.173.100.144:5060;user=phone
;tag=dc7-13c4-2401b7-46dea593-2401b7 Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441 CSeq: 118584736 BYE Contact: <sip:+1212381XXXX@66.197.142.69:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR" Content-Length: 0
On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists@avgs.ca>wrote:
Hi,
Digging a bit in mod_sofia releaved that it can be caused by a SIP code 482 (loop detected), 483 (too many hops) or 484 (address incomplete).
Do a SIP trace to sched more light on what's happening.
Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 m <mrene@avgs.ca>
_______________________________________________ FreeSWITCH-users mailing list 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