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

Thu Feb 18 21:44:19 CST 2016

On Thu, Feb 18, 2016 at 8:20 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:

> 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.
>
>
Oops.  I meant pjsip show endpoint.

> *CLI> pjsip show identifies
> No objects found.
>

This is the problem.  You should see something like...

Identify:  twilio-siptrunk-identify/twilio-siptrunk
      Match: 54.172.60.1/32
      Match: 54.172.60.3/32
      Match: 54.172.60.2/32
      Match: 54.172.60.0/32

If you use the uri_patterns then your config looks OK so watch Asterisk
when it starts to see if it prints any errors or warnings.

>
> 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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>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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