thr3ads.net - asterisk users - [asterisk-users] No matching endpoint found for incoming call from SIP trunk [Feb 2016]

If this information is useful, please help other people find it:
Share via:

Sonny Rajagopalan

2016-Feb-19 03:20 UTC

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

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
>-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/796c9e7f/attachment.html>

George Joseph

2016-Feb-19 03:44 UTC

head link

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

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.
>>>
>>> --
>>>
_____________________________________________________________________
>>> -- 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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
>-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/2ee9ae7e/attachment.html>

Sonny Rajagopalan

2016-Feb-19 04:24 UTC

head link

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

OK, I fixed it.  Here's what I did:

Well, I first saw a lot of errors  like this when Asterisk starts up (CLI
messages immediately upon startup):

[Feb 18 22:47:44] ERROR[5749]: netsock2.c:305 ast_sockaddr_resolve:
getaddrinfo("sillyapp.pstn.twilio.com;transport=tcp",
"(null)", ...): Name
or service not known
[Feb 18 22:47:44] ERROR[5749]: res_pjsip_endpoint_identifier_ip.c:186
ip_identify_match_handler: Address
'sillyapp.pstn.twilio.com;transport=tcp'
provided on ip endpoint identifier 'twilio-siptrunk-identify' did not
resolve to any address
[Feb 18 22:47:44] ERROR[5749]: config_options.c:720 aco_process_var: Error
parsing match=sillyapp.pstn.twilio.com;transport=tcp at line 0 of
[Feb 18 22:47:44] ERROR[5749]: res_pjsip_config_wizard.c:329 create_object:
Unable to apply object type 'identify' with id
'twilio-siptrunk-identify'.
Check preceeding errors.

However, I _am_ able to resolve them from the host (and yes, the ports to
twilio are open too):

$ nslookup sillyapp.pstn.twilio.com
Server:         172.31.0.2
Address:        172.31.0.2#53

Non-authoritative answer:
Name:   sillyapp.pstn.twilio.com
Address: 54.172.60.1
Name:   sillyapp.pstn.twilio.com
Address: 54.172.60.2
Name:   sillyapp.pstn.twilio.com
Address: 54.172.60.3
Name:   sillyapp.pstn.twilio.com
Address: 54.172.60.0

What I finally did to fix this is

[twilio-siptrunk]
type = wizard
sends_auth = yes
sends_registrations = no
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
outbound_auth/username = gobble
outbound_auth/password = degookdegook
endpoint/context = from-external
endpoint/disallow = all
endpoint/allow = ulaw
aor/qualify_frequency = 15


(Note, if you recall my earlier post/question on this list, I removed the
fix from that post (\;transport=tcp from remote_hosts) and stuck the fixes
you propose in *_uri_pattern etc.)

Now, I do see the identifies in pjsip show endpoint twilio-siptrunk:

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

And my incoming and outgoing calls via twilio work.

Phew!

Thanks again, George. You are a lifesaver!

On Thu, Feb 18, 2016 at 10:44 PM, George Joseph <george.joseph at
fairview5.com> wrote:
>
>
> 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.
>>>>
>>>> --
>>>>
_____________________________________________________________________
>>>> -- 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
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/321ec18e/attachment-0001.html>

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

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

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

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