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sonny.rajagopalan at g... Guest
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Posted: Mon Nov 16, 2015 8:36 pm Post subject: [asterisk-users] How do I enable instant messaging support f |
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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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engthyda at gmail.com Guest
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Posted: Mon Nov 16, 2015 9:17 pm Post subject: [asterisk-users] How do I enable instant messaging support f |
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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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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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sonny.rajagopalan at g... Guest
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Posted: Mon Nov 16, 2015 9:41 pm Post subject: [asterisk-users] How do I enable instant messaging support f |
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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@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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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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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_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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engthyda at gmail.com Guest
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Posted: Mon Nov 16, 2015 9:45 pm Post subject: [asterisk-users] How do I enable instant messaging support f |
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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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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.
--
_____________________________________________________________________
-- 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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--
_____________________________________________________________________
-- 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:
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sonny.rajagopalan at g... Guest
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Posted: Mon Nov 16, 2015 9:59 pm Post subject: [asterisk-users] How do I enable instant messaging support f |
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Thanks again. How do you create that message context in extensions.conf?
On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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.
--
_____________________________________________________________________
-- 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
|
--
_____________________________________________________________________
-- 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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engthyda at gmail.com Guest
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Posted: Tue Nov 17, 2015 2:04 am Post subject: [asterisk-users] How do I enable instant messaging support f |
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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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | Thanks again. How do you create that message context in extensions.conf?
On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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.
--
_____________________________________________________________________
-- 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
|
--
_____________________________________________________________________
-- 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:
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sonny.rajagopalan at g... Guest
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Posted: Tue Nov 17, 2015 7:52 am Post subject: [asterisk-users] How do I enable instant messaging support f |
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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@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | Thanks again. How do you create that message context in extensions.conf?
On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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@gmail.com (engthyda@gmail.com)> wrote:
Quote: | 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@gmail.com (sonny.rajagopalan@gmail.com)> wrote:
Quote: | 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.
--
_____________________________________________________________________
-- 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
|
--
_____________________________________________________________________
-- 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:
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