Sonny Rajagopalan
2016-Feb-19 02:25 UTC
[asterisk-users] 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/332b1144/attachment.html>
George Joseph
2016-Feb-19 02:56 UTC
[asterisk-users] No matching endpoint found for incoming call from SIP trunk
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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/ce9367bb/attachment.html>
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>