Sonny Rajagopalan
2016-Feb-17 13:38 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/3e2d4f46/attachment-0001.html>
Sonny Rajagopalan
2016-Feb-17 15:56 UTC
[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work
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 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 >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/8ad9f5ce/attachment.html>
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>
Maybe Matching Threads
- Asterisk 13.6.0/The simplest TCP configuration does not work
- No matching endpoint found for incoming call from SIP trunk
- Asterisk 13.6.0/The simplest TCP configuration does not work
- No matching endpoint found for incoming call from SIP trunk
- Asterisk 13.6.0/The simplest TCP configuration does not work