[jitsi-users] jingle call to asterisk


#1

Hi all,

I need to convert jingle calls from/to sip/pstn in my asterisk server.

On the asterisk cli, I see that the jabber module receives the call,
and in my dialplan I have the lines to answer it.
But my jitsi client remain indicating "connecting".

On ejabberd I see that both users are "online".
Most example are regarding communicating from/to google, so it seems that there is a difference between google and a local xmpp-server.

Anybody around here who did likewise? Suggestions?

Hans

···

______________________________________________________________________
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This message may contain information that is not intended for you. If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. The State accepts no liability for damage of any kind resulting from the risks inherent in the electronic transmission of messages.


#2

Ok I give .. Can someone please assist me How the heck do you remove
sent/received file attachments from the chat log from showing up every time
I start a new chat?

<img src=’/uploads/jitsi/original/1X/6d4ccb5e704d7717ec30d8f1b26a910d07db49b5.jpg’ width=‘684’ height=‘331’>

Thank you!

And yes I have "tried turning off the computer and turning it back on
again<http://www.youtube.com/watch?v=nn2FB1P_Mn8>
"

-Robert


#3

Hey Hans,

What version of Asterisk are you using? The currently stable version
only has experimental support for the Google version. It appears that
the latest development version (11 beta2) has a properly working Jingle
channel called chan_motif. You may want to give it a try.

Cheers,
Emil

···

On 18.09.12, 15:02, J.Witvliet@mindef.nl wrote:

Hi all,

I need to convert jingle calls from/to sip/pstn in my asterisk server.

On the asterisk cli, I see that the jabber module receives the call,
and in my dialplan I have the lines to answer it.
But my jitsi client remain indicating "connecting".

On ejabberd I see that both users are "online".
Most example are regarding communicating from/to google, so it seems that there is a difference between google and a local xmpp-server.

Anybody around here who did likewise? Suggestions?

Hans

______________________________________________________________________
Dit bericht kan informatie bevatten die niet voor u is bestemd. Indien u niet de geadresseerde bent of dit bericht abusievelijk aan u is toegezonden, wordt u verzocht dat aan de afzender te melden en het bericht te verwijderen. De Staat aanvaardt geen aansprakelijkheid voor schade, van welke aard ook, die verband houdt met risico's verbonden aan het elektronisch verzenden van berichten.

This message may contain information that is not intended for you. If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. The State accepts no liability for damage of any kind resulting from the risks inherent in the electronic transmission of messages.


#4

Hi Emil,

Yeah, I do follow the asterisk development,
And indeed they seems to be clearing things a bit up with chan_motif, although they are still discussing about changing config in a drastic way for asterisk_11.

Actually, I was using the older LTS-version, 1.8.15.1, with jabber.conf, jingle.conf, ....
Somehow I got the impression that with that older version it should be possible.

As I'm in favor of progress (as long as it adds something) and like to try something, I removed 1.8.15.1 and downloaded and built 11-beta1.
Edited (according to the asterisk11 wiki-page) extensions.conf, chan_motif.conf, jingle.conf and restarted.

Same behavior, except for minor details.
As soon as I start, ejabberd tells me that the defined user becomes online.

From jitsi I can send a text-message, which I see as I enabled "debug" in motif.conf

(This is actually progress, as in 1.8.15.1 I saw only empty strings coming along :wink:

But when starting an audio or an AV-call, I only see the xmpp-debug message (used to be jabber-debug-message).
Within de xmpp-message I see the capabilities (samplerate, codecs, address, port) from the jitsi-client.

Although I made a separate context in the dialplan, it seems never to get there: hence no answer :frowning:
Eventough I explicitly point to them in xmpp.conf and jingle.conf....
So my client remains in "connecting" for ever....

Hans

···

-----Original Message-----

From: Emil Ivov [mailto:emcho@jitsi.org]

Sent: Tuesday, September 18, 2012 4:34 PM
To: users@jitsi.java.net
Cc: Witvliet, J, CDC/IV/DCOPS/I&S/HIN
Subject: [jitsi-users] Re: jingle call to asterisk

Hey Hans,

What version of Asterisk are you using? The currently stable version
only has experimental support for the Google version. It appears that
the latest development version (11 beta2) has a properly working Jingle
channel called chan_motif. You may want to give it a try.

Cheers,
Emil

On 18.09.12, 15:02, J.Witvliet@mindef.nl wrote:

Hi all,

I need to convert jingle calls from/to sip/pstn in my asterisk server.

On the asterisk cli, I see that the jabber module receives the call,
and in my dialplan I have the lines to answer it.
But my jitsi client remain indicating "connecting".

On ejabberd I see that both users are "online".
Most example are regarding communicating from/to google, so it seems that there is a difference between google and a local xmpp-server.

Anybody around here who did likewise? Suggestions?

Hans

______________________________________________________________________
Dit bericht kan informatie bevatten die niet voor u is bestemd. Indien u niet de geadresseerde bent of dit bericht abusievelijk aan u is toegezonden, wordt u verzocht dat aan de afzender te melden en het bericht te verwijderen. De Staat aanvaardt geen aansprakelijkheid voor schade, van welke aard ook, die verband houdt met risico's verbonden aan het elektronisch verzenden van berichten.

This message may contain information that is not intended for you. If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. The State accepts no liability for damage of any kind resulting from the risks inherent in the electronic transmission of messages.


#5

Hey Hans,

What version of Asterisk are you using? The currently stable version
only has experimental support for the Google version. It appears that
the latest development version (11 beta2) has a properly working Jingle

That's actually 11 beta1. Sorry for the confusion.

Emil

···

On 18.09.12, 16:33, Emil Ivov wrote:

channel called chan_motif. You may want to give it a try.

Cheers,
Emil

On 18.09.12, 15:02, J.Witvliet@mindef.nl wrote:

Hi all,

I need to convert jingle calls from/to sip/pstn in my asterisk server.

On the asterisk cli, I see that the jabber module receives the call,
and in my dialplan I have the lines to answer it.
But my jitsi client remain indicating "connecting".

On ejabberd I see that both users are "online".
Most example are regarding communicating from/to google, so it seems that there is a difference between google and a local xmpp-server.

Anybody around here who did likewise? Suggestions?

Hans

______________________________________________________________________
Dit bericht kan informatie bevatten die niet voor u is bestemd. Indien u niet de geadresseerde bent of dit bericht abusievelijk aan u is toegezonden, wordt u verzocht dat aan de afzender te melden en het bericht te verwijderen. De Staat aanvaardt geen aansprakelijkheid voor schade, van welke aard ook, die verband houdt met risico's verbonden aan het elektronisch verzenden van berichten.

This message may contain information that is not intended for you. If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. The State accepts no liability for damage of any kind resulting from the risks inherent in the electronic transmission of messages.