[asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

Sonny Rajagopalan sonny.rajagopalan at gmail.com
Wed Feb 17 13:13:57 CST 2016


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 .
>>>>
>>>> --
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>>>
>>>
>>
>> --
>> _____________________________________________________________________
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>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>
>
> --
> _____________________________________________________________________
> -- 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
>
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> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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