[asterisk-dev] Asterisk 12 trunk setup

[Mark Michelson]

> Kilburn Abrahams <mailto:kilburna at gmail.com>
> Tuesday, December 31, 2013 5:31 PM
> Hi
>
> I am testing Asterisk 12 and got most things working, but cannot get a 
> trunk setup working correctly with the pjsip channel driver. The 
> provider provides IP security so no registering or credentials are 
> required.
>
> It complains about no route to host.
>
> 1.8 configs that works
>
> [maintrunk]
> type=peer
> host=1.2.3.4
> disallow=all
> allow=g729,alaw,ulaw
>
> and use Dial(SIP/maintrunk/${ARG1})
>
> V12 (does not work)
>
> [udpnonat]
> type=transport
> protocol=udp
> bind=0.0.0.0:5060
>
> [maintrunk]
> type=endpoint
> transport=udpnonat
> disallow=all
> allow=g729,alaw,ulaw
> aors=maintrunk
>
> [maintrunk]
> type=aor
> contact=sip:1.2.3.4:5060
>
> and use Dial(PJSIP/${ARG1}@maintrunk)
>
> It dials but does not connect to the provider. Is the config correct?
>
> Thank you for your time.
>
> /K
>
>
> ------------------------------------------------------------------------
Hi,

First off, thanks for the feedback. The configuration looks correct to 
me. I'm not 100% sure at what point during your outgoing call that the 
failure occurs, but I have a couple of suggestions:

1) Make sure that if you have loaded chan_sip.so and chan_pjsip.so that 
they are not both trying to listen on the same port. It may be that you 
are sending out your INVITE from chan_pjsip and then chan_sip is 
receiving the response from the provider. Of course, if we're failing 
even to send the INVITE out, then this is likely not the issue.

2) Run pjsip CLI commands ("pjsip list endpoints", "pjsip list aors", 
"pjsip show endpoints", and "pjsip show aors") to make sure that 
everything is as you expect it to be. It may be that there was some sort 
of trouble on startup reading your configuration, and so the endpoint or 
aor may not exist at all. If you notice that either the maintrunk 
endpoint or aor is not listed, check error and warning messages on 
startup to see what may have caused the object not to be created properly.

What error or warning messages are emitted on the CLI when the call 
failure occurs? At what point during the SIP INVITE transaction is the 
problem occurring?

Hopefully this is something that can be easily resolved.
Mark Michelson
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140102/ab81c3a3/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: compose-unknown-contact.jpg
Type: image/jpeg
Size: 770 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140102/ab81c3a3/attachment.jpg>