[asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".
jcolp at digium.com
Mon Jan 18 05:57:59 CST 2016
Jonathan H wrote:
> Would greatly appreciate any input into this currently-unanswered
> question on the forum:
> I posted it on Jan 6th, have tried so many things, so much forum/list
> searching and late nights since, but have had to admit defeat.
> Rather than duplicate it all here, I've posted my logs and conf files
> on that thread, too.
> Problem is that while there are quite a few sip examples, I have
> chosen to take the path of pjsip.
> Seems I can manage to attach Blink, Zoiper, Microsip and my ITSP with
> multiple extensions without problem to my Asterisk, but sip2sip has
> beaten me!
> It's presumably something ridiculously simple, but there comes a point
> where you can't see the wood for the trees.
> If someone can help me resolve this, I'll post a complete guide on
> Github Gist to help others in the future.
It is likely that the IP address that traffic is coming from differs
from the IP address resolved by res_pjsip_endpoint_identifier_ip.
Currently that module is dumb and just does an A record lookup, it does
not do any SRV or NAPTR lookup (which sip2sip likely uses). As a result
when the INVITE comes in it does not identify it. You will need to
determine the possible IP addresses and create your own identify section
to match on them as the correct endpoint (I don't use wizards so don't
know how to configure it with them).
The current IP addresses possible being the following:
proxy.sipthor.net. 60 IN A 22.214.171.124
proxy.sipthor.net. 60 IN A 126.96.36.199
proxy.sipthor.net. 60 IN A 188.8.131.52
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
More information about the asterisk-users