[asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

[Jonathan H]

Gaaaah! 15 minutes after reading your answer, I had it working perfectly!

Thank you!

Before I type it up, here's what works for me - can you see any
obvious flaws or hidden dangers here?

-----------------------------------------------------------------------------
pjsip.conf
-----------------------------------------------------------------------------

[acl]
type = acl
deny = 0.0.0.0/0.0.0.0
permit = 81.23.228.129,81.23.228.150,85.17.186.7

-----------------------------------------------------------------------------
pjsip_wizard.conf
-----------------------------------------------------------------------------

[sip2sipusername]
type = wizard
sends_auth = yes
sends_registrations = yes
remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info
outbound_auth/username = sip2sipusername
outbound_auth/password = sip2sippassword
endpoint/allow = alaw
endpoint/context = sip2sipusername
registration/contact_user = sip2sipusername
outbound_proxy = proxy.sipthor.net
endpoint/language=en_GB

-----------------------------------------------------------------------------
extensions.conf
-----------------------------------------------------------------------------

[sip2sipusername]
exten => sip2sipusername,1,NoOp()
same => n, playback(hello-world)

-----------------------------------------------------------------------------

Again, thank you so much. I wish I'd discovered this mailing list
weeks ago - I'd assumed it was like those mailing lists that were "in
sync" with the forum.

Lesson learnt, and again, thank you.

On 18 January 2016 at 11:57, Joshua Colp <jcolp at digium.com> wrote:
> Jonathan H wrote:
>>
>> Would greatly appreciate any input into this currently-unanswered
>> question on the forum:
>>
>> http://forums.asterisk.org/viewtopic.php?f=1&t=96496
>>
>> 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       81.23.228.129
> proxy.sipthor.net.      60      IN      A       85.17.186.7
> proxy.sipthor.net.      60      IN      A       81.23.228.150
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
> --
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