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[sonny.rajagopalan at g...]

OK. I will report with my findings. It appears increasingly likely that I have done something very silly on my side. It is a little perplexing that the EXACT setup (on the same machine) worked for UDP ...

On Wed, Feb 17, 2016 at 8:23 AM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:

[sonny.rajagopalan at g...]

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.


On Wed, Feb 17, 2016 at 8:28 AM, Sonny Rajagopalan <sonny.rajagopalan@gmail.com (sonny.rajagopalan@gmail.com)> wrote:

[jcolp at digium.com]

Sonny Rajagopalan wrote:

[asterisk_list at earth...]

On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote:

[sonny.rajagopalan at g...]

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@earthshod.co.uk (asterisk_list@earthshod.co.uk)> wrote:

[sonny.rajagopalan at g...]

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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:

[george.joseph at fairv...]

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan <sonny.rajagopalan@gmail.com (sonny.rajagopalan@gmail.com)> wrote:

[sonny.rajagopalan at g...]

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

[george.joseph at fairv...]

On Wed, Feb 17, 2016 at 12:13 PM, Sonny Rajagopalan <sonny.rajagopalan@gmail.com (sonny.rajagopalan@gmail.com)> wrote:

[sonny.rajagopalan at g...]

George,

May I propose we improve the documentation on the Asterisk Wiki? I thought I would have spent far less time here (though you folks have been mightily helpful, and thanks again!) should the documentation for the TCP transport be improved in both the Wiki and specifically, in ${ASTERISK_HOME}/configs/samples/, bundled as part of the code. I want to see Asterisk as a product succeed (even more) and Asterisk in its new version succeed wildly. 


I don't know if you folks allow outside developers to pitch in, but depending on a number of factors, I might contribute to https://gerrit.asterisk.org/ if that is within Asterisk's policy. Again, depending on a number of factors, including legal.


Here, specifically, is the list of improvements I propose:


(a) One full example showing how a TCP based Asterisk platform should work in the PJSIP world, including both SIP over TCP-compliant SIP trunk configuration using pjsip_wizard.conf
(b) One complete example reflected in the distributed code samples within Asterisk code, in ${ASTERISK_HOME}/configs/samples/.
(c) A full SIP trace for Asterisk, the working examples, for all manner of transports (UDP, TCP, TLS), for REGISTER, INVITE etc.


Hope this helps.

Thanks again!


On Wed, Feb 17, 2016 at 3:48 PM, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:

[jcolp at digium.com]

Sonny Rajagopalan wrote: