[asterisk-users] No matching endpoint found for incoming call from SIP trunk

[Sonny Rajagopalan]

Thanks George, for your mighty quick response.

I made the changes (re: server_uri_pattern etc.) and still, no luck--it
fails for the same error.

BTW, there is nothing for transport (but this is the same config from my
SIP/UDP + Twilio days, which worked):

*CLI> pjsip show transport twilio-siptrunk
Unable to find object twilio-siptrunk.

*CLI> pjsip show identifies
No objects found.

I did add ;transport=tcp to my Origination URI after wireshark revealed
everything was received as UDP into Asterisk, so we can rule out that issue
(I confirmed that I am getting TCP based SIP INVITEs from Twilio, and
confirmed that the Asterisk server sends a 401 Unauthorized for the
initiation INVITE).

Per the pjsip_wizard.conf samples, long ago, I removed pjsip.conf-based
Twilio config and placed it all in pjsip_wizard.conf.

Thanks, re: wiki, I will be using it heavily, for sure ;-)

On Thu, Feb 18, 2016 at 9:56 PM, George Joseph <george.joseph at fairview5.com>
wrote:

>
>
> On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> Hello,
>>
>> I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio
>> gateway. I am able to make calls outbound through the gateway, but I am not
>> able to make calls into the PBX from external PSTN.
>>
>> Specifically, an incoming call is _received_ by Asterisk, but it is not
>> able to route the call internally owing to the following error:
>>
>> [Feb 18 21:08:47] NOTICE[4606]: res_pjsip/pjsip_distributor.c:347
>> log_unidentified_request: Request from '<
>> sip:+19725551212 at sillyapp.pstn.twilio.com;isup-oli=62;pstn-params=808481808882;cpc=ordinary>'
>> failed for '11.12.13.14:38124' (callid:
>> 3532ca0d142e6ce92f0259fd51cb5e43 at 0.0.0.0) - No matching endpoint found
>>
>> The last time I had this error, I was dealing with another SIP trunk and
>> the issue was that I had mixed up "identify" and with "identity", but I
>> have not such type in my pjsip_wizard.conf which looks like this:
>>
>> type = wizard
>> sends_auth = yes
>> sends_registrations = no
>> remote_hosts = sillyapp.pstn.twilio.com\;transport=tcp
>>
>
> ​I'll bet that if you do a "pjsip show transport twilio"​ you won't see
> any Identify or Matches.  I think there's a bug in the wizard that's not
> correctly handling the "\;transport=tcp" in all cases when it's appended to
> remote_hosts.  I'll check on it tomorrow.
>
> ​Do this instead:​
>
> remote_hosts = sillyapp.pstn.twilio.com
> server_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP
> client_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP
> contact_pattern = sip:${REMOTE_HOST}\;transport=TCP
>
> Also, make sure that your Twilio "Origination URI" has the ";transport=tcp"
> appended.
>
> ​I'll be working ​on the wiki tomorrow as well. :)
>
>
>
>> outbound_auth/username = gobble
>> outbound_auth/password = degookdegook
>> endpoint/context = from-external
>> endpoint/disallow = all
>> endpoint/allow = ulaw
>> aor/qualify_frequency = 15
>>
>> And--of course, I do have the DID configured on my extension, and in the
>> dialplan "from-external" (confirmed using dialplan show from-external).
>>
>> What is incorrect, and what should I be doing?
>>
>> Any help is appreciated deeply.
>>
>> Thank you,
>>
>> Cheers,
>> Sonny.
>>
>> --
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
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