We’‘re very excited to announce the first release of Asterisk-IM – a server plugin the integrates the Asterisk phone system and Jive Messenger instant messaging. We’‘ve been using the module internally for almost a month and it’'s very powerful (and convenient) to have unified presence between the phone and IM. Several commercial instant messaging servers have announced a similar integration with commercial phone systems, but Asterisk-IM is the first fully Open Source stack to achieve the same feature set.
There are still a few rough edges in the 1.0 beta 1 release, but we plan to move quickly towards a final 1.0 release. We have many ideas for future features as well.
Can you provide a simple overview of how exactly each feature of Asterisk-IM works?
For example, how does it know when a user is on the phone, and how does it “signal” that info to your jabber buddies.
Also how do “Call Notification” and “Originate Calls” features work? What messages are sent to/from the jabber client and how does the Asterisk-IM plugin dial the number??
The Asterisk Manager Proxy is a simple TCP protocol that provides away to receive notifications from asterisk (events) and have asterisk perform tasks (actions).
“Call Notification” is performed by sending a message packet with phone-event packet extension to the client when the asterisk sends the plugin a phone event.
“Originate Calls” is performed when a client sends the Asterisk-IM component a IQ packet with an extension of phone-action.
For more details on actions and events see the Phone Integration proto-JEP:
I just upgraded our CallManagers to the latest revision this past week (4.1.3), and my boss jokingly said, “Great job. Now all you need to do is figure out how to integrate that with our Jabber clients.” He was joking, but I started wondering if that would be possible. And then you guys announce this Asterisk-IM plug-in and prove that it is. Perfect timing!!
So, this will be my secret project for the next few months.
I’‘m a novice Java programmer, but I will definately help out in any way I can. Feel free to PM me or catch me on Jive’‘s XMPP server (user “hrothgar”) if you have any questions about Cisco’'s implementation or need help with Cisco docs.
I’'ll try to get time to look over your server-side implementation and read up on Cisco docs:
… it looks like JTAPI could be used to integrate with a number of phone systems. That would be quite cool.
Wow, I was excited to see Jabber and asterisk together in the same conversation. I’'ve been a long time fan of both and was exploring options to integrate the two, then BAM… you guys spring this on me and I cant wait to finish the rebuild of my asterisk box, now your saying that there might be options for other phone systems… I must be dreaming, I have 2 clients with nortel VOIP and Jabber who would be in heaven if they could tie the two together.
I’'m all ears and hope I can help were ever possible…
The best place to start is to build a kick-butt Asterisk integration and then expand from there. What we need to do to make that happen:
Get XMPP client authors to build in support for the protocol so that people can start using the integration to its fullest.
Bring in Asterisk experts to the project so that we can figure out how to implement more advanced features likes turning a group chat into a group conference call.
Get XMPP client authors to build in support for the protocol so that people can start using the integration to its fullest.[/i]
I’‘ve sent an email to the PSI team to bring this new integration to their attention, I have a friend who is on good terms with the guys a NeosMT and I’'ll have him fire them an email to give them the heads up.
Bring in Asterisk experts to the project so that we can figure out how to implement more advanced features likes turning a group chat into a group conference call.[/i]
I’‘ve posted messages to both the Asterisk Users and Dev mailing list to let people know about the new project and I’'ve also updated the Voip-Info wiki to list Jive Messenger and Asterisk-IM as Third part software. http://www.voip-info.org/tiki-index.php?page=Asterisk
I hope this gets the ball rolling…
Peter
PS. If credit is due, Is there someones name in particular I should reference or was this a Jive original?