[asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?

Tue Nov 17 06:51:24 CST 2015

I did this, but I do not see this as part of my dialplan. And the chat
still does not work. Why is this not documented anywhere for Asterisk 13?
Is this supported still, in Asterisk 13.x?

On Tue, Nov 17, 2015 at 2:03 AM, Thyda ENG <engthyda at gmail.com> wrote:

> Here is my extension .
> [astsms]
> exten => _.,1,NoOp(SMS receiving dialplan invoked)
> exten => _.,n,NoOp(To ${MESSAGE(to)})
> exten => _.,n,NoOp(From ${MESSAGE(from)})
> exten => _.,n,NoOp(Body ${MESSAGE(body)})
> exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
> exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
> exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
> exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" !=
> "SUCCESS"]?sendfailedmsg)
> exten => _.,n,Hangup()
>
> exten =>
> _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}]
> Your message to ${EXTEN} has failed. Retry later.")
> exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
> exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
> exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
> exten => _.,n,Hangup()
> exten => _.,n,Hangup()
>
> On Tue, Nov 17, 2015 at 9:58 AM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> Thanks again. How do you create that message context in extensions.conf?
>>
>> On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda at gmail.com> wrote:
>>
>>> According to what I have done , I add the message_context to the
>>> pjsip.endpoint_custom.conf in /etc/asterisk and then I create that
>>> message_context in the extension.conf, and it works.
>>>
>>> On Tue, Nov 17, 2015 at 9:34 AM, Sonny Rajagopalan <
>>> sonny.rajagopalan at gmail.com> wrote:
>>>
>>>> So, the only thing that is needed in the endpoint definition in
>>>> pjsip.conf (there is no such file pjsip.endpoint_custom.conf) is
>>>>
>>>> *message_context=astsms*
>>>>
>>>> Is that correct? Anything I need to do in extensions.conf? I see that
>>>> the messages are received at Asterisk (when I turn on pjsip set logger on)
>>>> but they are not delivered to the other endpoint. What gives?
>>>>
>>>> Any help appreciated. Thanks!
>>>>
>>>> On Mon, Nov 16, 2015 at 9:16 PM, Thyda ENG <engthyda at gmail.com> wrote:
>>>>
>>>>> The default message context for the pjsip is the same the call
>>>>> context, so to set the new message context for the pjsqip you need to
>>>>> modify your pjsip.endpoint_custom.conf and add the message context as in
>>>>> the example below :
>>>>>
>>>>> [100]
>>>>>
>>>>> type=endpoint
>>>>>
>>>>> aors=100
>>>>>
>>>>> auth=100-auth
>>>>>
>>>>> allow=ulaw,alaw,gsm,g726
>>>>>
>>>>> context=from-internal
>>>>>
>>>>> callerid=device <100>
>>>>>
>>>>> dtmf_mode=rfc4733
>>>>>
>>>>> use_avpf=no
>>>>>
>>>>> ice_support=no
>>>>>
>>>>> media_use_received_transport=no
>>>>>
>>>>> trust_id_inbound=yes
>>>>>
>>>>> media_encryption=no
>>>>>
>>>>> rtp_symmetric=yes
>>>>>
>>>>> rewrite_contact=yes
>>>>>
>>>>> *message_context=astsms*
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan <
>>>>> sonny.rajagopalan at gmail.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> I am looking for documentation support for enabling instant messaging
>>>>>> between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as
>>>>>> Zoiper. Where do I enable this support on the server side and does it need
>>>>>> anything on the client side? I see plenty of online help for chan_sip, but
>>>>>> nothing for chan_pjsip.
>>>>>>
>>>>>> I imagine there is both pjsip.conf configuration and extensions.conf
>>>>>> configuration?
>>>>>>
>>>>>> Any help is appreciated.
>>>>>>
>>>>>> Thanks,
>>>>>> Sonny.
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
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>>>>>
>>>>>
>>>>> --
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>>>>
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>>>
>>>
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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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>>    http://lists.digium.com/mailman/listinfo/asterisk-users
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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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