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


 
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sonny.rajagopalan at g...
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PostPosted: Thu Feb 18, 2016 9:25 pm    Post subject: [asterisk-users] No matching endpoint found for incoming cal Reply with quote

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@sillyapp.pstn.twilio.com ([email]sip%3A%2B19725551212@sillyapp.pstn.twilio.com[/email]);isup-oli=62;pstn-params=808481808882;cpc=ordinary>' failed for '3532ca0d142e6ce92f0259fd51cb5e43@0.0.0.0 (3532ca0d142e6ce92f0259fd51cb5e43@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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george.joseph at fairv...
Guest





PostPosted: Thu Feb 18, 2016 9:56 pm    Post subject: [asterisk-users] No matching endpoint found for incoming cal Reply with quote

On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan <sonny.rajagopalan@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote:
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@sillyapp.pstn.twilio.com ([email]sip%3A%2B19725551212@sillyapp.pstn.twilio.com[/email]);isup-oli=62;pstn-params=808481808882;cpc=ordinary>' failed for '3532ca0d142e6ce92f0259fd51cb5e43@0.0.0.0 (3532ca0d142e6ce92f0259fd51cb5e43@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. Smile


 
Quote:
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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New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

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sonny.rajagopalan at g...
Guest





PostPosted: Thu Feb 18, 2016 10:21 pm    Post subject: [asterisk-users] No matching endpoint found for incoming cal Reply with quote

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 Wink


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



On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan <sonny.rajagopalan@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote:
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@sillyapp.pstn.twilio.com ([email]sip%3A%2B19725551212@sillyapp.pstn.twilio.com[/email]);isup-oli=62;pstn-params=808481808882;cpc=ordinary>' failed for '3532ca0d142e6ce92f0259fd51cb5e43@0.0.0.0 (3532ca0d142e6ce92f0259fd51cb5e43@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. Smile


 
Quote:
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

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george.joseph at fairv...
Guest





PostPosted: Thu Feb 18, 2016 10:45 pm    Post subject: [asterisk-users] No matching endpoint found for incoming cal Reply with quote

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


 

Quote:

*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.

 

Quote:


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 Wink


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



On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan <sonny.rajagopalan@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote:
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@sillyapp.pstn.twilio.com ([email]sip%3A%2B19725551212@sillyapp.pstn.twilio.com[/email]);isup-oli=62;pstn-params=808481808882;cpc=ordinary>' failed for '3532ca0d142e6ce92f0259fd51cb5e43@0.0.0.0 (3532ca0d142e6ce92f0259fd51cb5e43@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. Smile


 
Quote:
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:
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sonny.rajagopalan at g...
Guest





PostPosted: Thu Feb 18, 2016 11:24 pm    Post subject: [asterisk-users] No matching endpoint found for incoming cal Reply with quote

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@fairview5.com (george.joseph@fairview5.com)> wrote:
Quote:



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


 

Quote:

*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.

 

Quote:


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 Wink


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



On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan <sonny.rajagopalan@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote:
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@sillyapp.pstn.twilio.com ([email]sip%3A%2B19725551212@sillyapp.pstn.twilio.com[/email]);isup-oli=62;pstn-params=808481808882;cpc=ordinary>' failed for '3532ca0d142e6ce92f0259fd51cb5e43@0.0.0.0 (3532ca0d142e6ce92f0259fd51cb5e43@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. Smile


 
Quote:
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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_____________________________________________________________________
-- 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
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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