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

Sun Oct 18 07:25:10 PDT 2009

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 at 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 at 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 at 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
>> mrene at avgs.ca
>>
>
>
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