[asterisk-users] No matching endpoint found for incoming call from SIP trunk
Sonny Rajagopalan
sonny.rajagopalan at gmail.com
Thu Feb 18 22:24:16 CST 2016
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.
>>>>
>>>> --
>>>> _____________________________________________________________________
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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
>>
>
>
> --
> _____________________________________________________________________
> -- 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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