Well, if it's inside the dialplan you'll see it in the log. You may want to rotate logs frequently while tracking this down. It might also help if you enable uuid logging so that you can filter down to just a single call's log lines. You may also wish to examine the XML CDRs for these calls. I'd be curious to see what's in the call flows. -MC On Wed, Jun 1, 2011 at 2:38 PM, Bill W. <freeswitch@aastral.net> wrote:

Hey Anthony,

Thanks very much for your input. Can you recommend a way I can prove that to myself? Is there some sort of logging to turn on to see that?

The reason I ask is that for this instance I receive a call in on one sofia profile and then use the bridge command to send it out another sofia profile. I don't use the transfer application to complete the call.

On 6/1/11 4:36 PM, Anthony Minessale wrote:

...

routing loop, its transferring more that 73 times inside the dialplan

On Wed, Jun 1, 2011 at 3:10 PM, Bill W. <freeswitch@aastral.net> wrote:

...

Hey All,

About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR failure.

I've enabled debugging, including logging SIP messages, and can't seem to find the cause of the error.

Freeswitch goes from setting variables to exchange routing error: switch_channel.c:935 EXPORTING[export_vars] to switch_ivr_originate.c:2447 Cannot create outgoing channel of type [sofia] cause: [EXCHANGE_ROUTING_ERROR]

Whenever that happens, "switch_channel.c:816 New Channel" is never called and I don't see any outbound SIP messages to the IP in question.

So if no new channel is created, how can there be a routing error?

Any insight or advice on how to debug this would be greatly appreciated.

Thanks!

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its not only transfers, its possibly a call looping in a circle to the same box. Are you using your own domain in a bridge app somewhere for instance: <action application="bridge" data="sofia/internal/1234@domain.com"/> And if your own box is domain.com and you mistakenly are trying to call the extension registered to 1234 you would actually call yourself over sip in a loop up to 73 times until it hungup with that cause. On Thu, Jun 2, 2011 at 8:29 AM, Bill W. <freeswitch@aastral.net> wrote:

Hey Michael,

Thanks so much for the advice.  I've already enabled uuid logging, and have grepped the log for the uuid of the call that failed.  As in my initial email, it just goes from 'setting variables' to the error.  No new channel is created.

There are no transfers in the dialplan nor in freeswitch.log.

I don't have XML CDRs enabled, but the cdr-csv  log entry is unremarkable.

I'll enable XML CDR and see if that can provide some more information.

On 6/1/11 5:46 PM, Michael Collins wrote:

...

Well, if it's inside the dialplan you'll see it in the log. You may want to rotate logs frequently while tracking this down. It might also help if you enable uuid logging so that you can filter down to just a single call's log lines. You may also wish to examine the XML CDRs for these calls. I'd be curious to see what's in the call flows.

-MC

On Wed, Jun 1, 2011 at 2:38 PM, Bill W. <freeswitch@aastral.net <mailto:freeswitch@aastral.net>> wrote:

    Hey Anthony,

    Thanks very much for your input.  Can you recommend a way I can prove     that to myself?  Is there some sort of logging to turn on to see that?

    The reason I ask is that for this instance I receive a call in on one     sofia profile and then use the bridge command to send it out another     sofia profile. I don't use the transfer application to complete the     call.

    On 6/1/11 4:36 PM, Anthony Minessale wrote:     > routing loop, its transferring more that 73 times inside the dialplan     >     > On Wed, Jun 1, 2011 at 3:10 PM, Bill W. <freeswitch@aastral.net     <mailto:freeswitch@aastral.net>> wrote:     >> Hey All,     >>     >> About once or twice a day, I'm getting an EXCHANGE_ROUTING_ERROR     failure.     >>     >> I've enabled debugging, including logging SIP messages, and can't     seem     >> to find the cause of the error.     >>     >>     >> Freeswitch goes from setting variables to exchange routing error:     >> switch_channel.c:935 EXPORTING[export_vars]     >> to     >> switch_ivr_originate.c:2447 Cannot create outgoing channel of type     >> [sofia] cause: [EXCHANGE_ROUTING_ERROR]     >>     >>     >> Whenever that happens, "switch_channel.c:816 New Channel" is never     >> called and I don't see any outbound SIP messages to the IP in     question.     >>     >> So if no new channel is created, how can there be a routing error?     >>     >> Any insight or advice on how to debug this would be greatly     appreciated.     >>     >> Thanks!     >>     >>     >>     >> _______________________________________________     >> FreeSWITCH-users mailing list     >> FreeSWITCH-users@lists.freeswitch.org     <mailto: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     >>     >     >     >

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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

then look for a low max-forwards header in the sip traffic On Thu, Jun 2, 2011 at 11:07 AM, Bill W. <freeswitch@aastral.net> wrote:

Hey Anthony,

All the bridge commands are of the form: sofia/profile/18005551212@1.2.3.4

Also, if it was looping, would I see that in the logs?  It would parse the dialplan 73 times, correct?  Because I don't see that.

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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900

Thanks for the followup email. It's always nice to know how these problems eventually get solved... -MC On Mon, Jun 6, 2011 at 11:24 AM, Bill W. <freeswitch@aastral.net> wrote:

Hey Anthony/Michael,

After a lot of digging, it turns out the problem was the Max-Forwards on the incoming SIP traffic was too low for FreeSWITCH to complete the call.

Thanks so much for all your help!

On 6/2/11 1:13 PM, Anthony Minessale wrote:

...

then look for a low max-forwards header in the sip traffic

On Thu, Jun 2, 2011 at 11:07 AM, Bill W. <freeswitch@aastral.net> wrote:

...

Hey Anthony,

All the bridge commands are of the form: sofia/profile/18005551212@1.2.3.4

Also, if it was looping, would I see that in the logs? It would parse the dialplan 73 times, correct? Because I don't see that.

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