asterisk users - Feb 2016 - No matching endpoint found for incoming call from SIP trunk

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
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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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.
>
> --
> _____________________________________________________________________
> -- 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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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.
>>
>> --
>> _____________________________________________________________________
>> -- 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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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