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<font color="#9FA2A5"><span style="padding-left:6px">Tuesday, December
31, 2013 5:31 PM</span></font></div></div></div>
<div style="color:#888888;margin-left:24px;margin-right:24px;"
__pbrmquotes="true" class="__pbConvBody"><div>Hi<br><br>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. <br><br>It
complains about no route to host. <br><br>1.8 configs that works<br><br>[maintrunk]<br>type=peer<br>host=1.2.3.4<br>disallow=all<br>allow=g729,alaw,ulaw<br><br>and
use Dial(SIP/maintrunk/${ARG1})<br><br>V12 (does not work) <br><br>[udpnonat]<br>type=transport<br>protocol=udp<br>bind=0.0.0.0:5060<br><br>[maintrunk]<br>type=endpoint<br>transport=udpnonat<br>disallow=all<br>allow=g729,alaw,ulaw<br>aors=maintrunk<br><br>[maintrunk]<br>type=aor<br>contact=<a class="moz-txt-link-freetext" href="sip:1.2.3.4:5060">sip:1.2.3.4:5060</a><br><br>and
use Dial(PJSIP/${ARG1}@maintrunk)<br><br>It dials but does not connect
to the provider. Is the config correct?<br><br>Thank you for your time.<br><br>/K<br><br><br></div><hr
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Hi, <br>
<br>
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:<br>
<br>
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.<br>
<br>
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.<br>
<br>
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?<br>
<br>
Hopefully this is something that can be easily resolved.<br>
Mark Michelson<br>
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