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.jos...@fairview5.com > wrote: > > > On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < > sonny.rajagopa...@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.rajagopa...@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_l...@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 >
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