George Joseph
2016-Feb-17 17:43 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote:> I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am able to make > "from-internal" context calls (i.e., calls that do not leave the VoIP > island). > > Here's what I have right now in pjsip_wizard.conf (again, I have removed > it from /etc/asterisk/ because Asterisk won't even work for "from-internal" > calls with the conf in /etc/asterisk) > > [twilio-siptrunk] > type = wizard > sends_auth = yes > sends_registrations = no > remote_hosts = silly.pstn.twilio.com >remote_hosts = silly.pstn.twilio.com ?\;transport=TCP? outbound_auth/username = username> outbound_auth/password = sillypassword > endpoint/context = from-external ;;; change later > endpoint/disallow = all ;;; change later > endpoint/allow = ulaw ;;; change later > aor/qualify_frequency = 15 > > What should I change/add/modify above to make Asterisk and Twilio work > with TCP? Note that I do not have to trigger a use of the twilio sip trunk > for my Asterisk daemon to not work for TCP. If I have the pjsip_wizard in > /etc/asterisk, it does not work for _any_ call, regardless of whether or > not the call should use the Twilio SIP trunk. > > (again, the same asterisk configuration on the same machine connected to > the same twilio SIP trunk worked for UDP) > > If anyone knows the trick to make pjsip_wizard.conf work with twilio, I > would very much appreciate any insight... > > Thanks, > Sonny. > > On Wed, Feb 17, 2016 at 8:38 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Yes, it is enabled on port 5060. I do receive a TCP ACK back from the >> server, so I know the TCP segment is received at the server hosting the >> Asterisk build. >> >> On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles < >> asterisk_list at earthshod.co.uk> wrote: >> >>> On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: >>> > OK. Let me ask this. Is anything else necessary, except choosing TCP >>> as the >>> > preferred protocol on the client, to make TCP w Asterisk work? At the >>> > moment, I have only changed one line in pjsip.conf from my working UDP >>> > setup: >>> > >>> > [transport-tcp] >>> > type=transport >>> > protocol=tcp ; <--------------- only this line was changed. >>> >>> Presumably you have firewall rules in action. Did you enable TCP on port >>> 5060? >>> >>> -- >>> AJS >>> >>> Note: Originating address only accepts e-mail from list! If replying >>> off- >>> list, change address to asterisk1list at earthshod dot co dot uk . >>> >>> -- >>> _____________________________________________________________________ >>> -- 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/20160217/403eae9a/attachment.html>
Sonny Rajagopalan
2016-Feb-17 19:13 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
Wow. Incredible. That worked. The backslash is important there; I kept trying with no backslash and followed the instructions in pjsip_wizard.conf.sample (in configs/samples) and it says we have to say transport=tcp ; the only example however talks about ipv4. Is this documented somewhere and I just missed it?? So, let me sum the issues and their solutions: (a) Inside/from-internal calling. Only need transport=tcp in pjsip.conf. No need to update every SIP (user) endpoint's transport, though that did not disrupt anything. (b) For pjsip_wizard configuration, add the transport into the remote_hosts line like so noting that the backslash is important otherwise the transport part of the line is a comment! remote_hosts = silly.pstn.twilio.com?\;transport=tcp Simple errors, but vexing, vexing, vexing issues. Thanks, George, and thanks Joshua, for your time! On Wed, Feb 17, 2016 at 12:43 PM, George Joseph <george.joseph at fairview5.com> wrote:> > > On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I made some progress. The first thing I have realized is that it is my >> Twilio configuration in pjsip_wizard.conf that was killing me. I have since >> removed that entire file from /etc/asterisk and I am able to make >> "from-internal" context calls (i.e., calls that do not leave the VoIP >> island). >> >> Here's what I have right now in pjsip_wizard.conf (again, I have removed >> it from /etc/asterisk/ because Asterisk won't even work for "from-internal" >> calls with the conf in /etc/asterisk) >> >> [twilio-siptrunk] >> type = wizard >> sends_auth = yes >> sends_registrations = no >> remote_hosts = silly.pstn.twilio.com >> > > remote_hosts = silly.pstn.twilio.com > ?\;transport=TCP? > > > outbound_auth/username = username >> outbound_auth/password = sillypassword >> endpoint/context = from-external ;;; change later >> endpoint/disallow = all ;;; change later >> endpoint/allow = ulaw ;;; change later >> aor/qualify_frequency = 15 >> >> What should I change/add/modify above to make Asterisk and Twilio work >> with TCP? Note that I do not have to trigger a use of the twilio sip trunk >> for my Asterisk daemon to not work for TCP. If I have the pjsip_wizard in >> /etc/asterisk, it does not work for _any_ call, regardless of whether or >> not the call should use the Twilio SIP trunk. >> >> (again, the same asterisk configuration on the same machine connected to >> the same twilio SIP trunk worked for UDP) >> >> If anyone knows the trick to make pjsip_wizard.conf work with twilio, I >> would very much appreciate any insight... >> >> Thanks, >> Sonny. >> >> On Wed, Feb 17, 2016 at 8:38 AM, Sonny Rajagopalan < >> sonny.rajagopalan at gmail.com> wrote: >> >>> Yes, it is enabled on port 5060. I do receive a TCP ACK back from the >>> server, so I know the TCP segment is received at the server hosting the >>> Asterisk build. >>> >>> On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles < >>> asterisk_list at earthshod.co.uk> wrote: >>> >>>> On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: >>>> > OK. Let me ask this. Is anything else necessary, except choosing TCP >>>> as the >>>> > preferred protocol on the client, to make TCP w Asterisk work? At the >>>> > moment, I have only changed one line in pjsip.conf from my working UDP >>>> > setup: >>>> > >>>> > [transport-tcp] >>>> > type=transport >>>> > protocol=tcp ; <--------------- only this line was changed. >>>> >>>> Presumably you have firewall rules in action. Did you enable TCP on >>>> port 5060? >>>> >>>> -- >>>> AJS >>>> >>>> Note: Originating address only accepts e-mail from list! If replying >>>> off- >>>> list, change address to asterisk1list at earthshod dot co dot uk . >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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/20160217/5925e2b0/attachment.html>
George Joseph
2016-Feb-17 20:48 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote:> Wow. Incredible. That worked. The backslash is important there; I kept > trying with no backslash and followed the instructions in > pjsip_wizard.conf.sample (in configs/samples) and it says we have to say > > transport=tcp ; the only example however talks about ipv4. > > Is this documented somewhere and I just missed it?? > > So, let me sum the issues and their solutions: > > (a) Inside/from-internal calling. Only need transport=tcp in pjsip.conf. > No need to update every SIP (user) endpoint's transport, though that did > not disrupt anything. > (b) For pjsip_wizard configuration, add the transport into the > remote_hosts line like so noting that the backslash is important otherwise > the transport part of the line is a comment! > > remote_hosts = silly.pstn.twilio.com?\;transport=tcp > > Simple errors, but vexing, vexing, vexing issues. >One thing to be aware of...? There is currently a PJSIP bug when using TCP and TLS that shows up if you explicitly set transport= on an endpoint (or in the wizard). It's best to leave transport unset and let PJSIP determine the transport from the ;transport= parameter of the URI.>From a wizard perspective, if you have lots of TCP or TLS endpoints, use atemplate like so... [tcp-template](!) server_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP client_uri_pattern = sip:${REMOTE_HOST}\;transport=TCP contact_pattern = sip:${REMOTE_HOST}\;transport=TCP [tls-template](!) server_uri_pattern = sips:${REMOTE_HOST}\;transport=TLS client_uri_pattern = sips:${REMOTE_HOST}\;transport=TLS contact_pattern = sips:${REMOTE_HOST}\;transport=TLS [tcp-provider](tcp-template] remote_hosts = my.provider.net Let me know if the wiki can use some clarification. I haven't updated it in a while.> > Thanks, George, and thanks Joshua, for your time! > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/0b73c7b7/attachment.html>
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