[Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?

Michael Jerris mike at jerris.com
Sun Oct 18 18:56:05 PDT 2009


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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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
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