asterisk users - Jan 2016 - Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

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.

Thanks.

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

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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