Re: OGo and Asterisk integration

Helge Hess <[email protected]>
Newsgroups gmane.comp.cms.opengroupware.discuss.general
Message-ID <[email protected]>
On Sep 27, 2004, at 22:23, John Helmuth wrote:
> It looks like there's 2 simple ways to get Asterix to dial out.

Great resource lookup!

> The first is to drop a text file detailing the call (a call file) into 
> a spool directory.  The file contains information about the 2 ends of 
> the call.   http://www.voip-info.org/wiki-Asterisk+auto-dial+out

Yup, should be trivial to implement as a CTI plugin. Please attach 
additional information (eg the exact file format) to the enh-request:
   http://bugzilla.opengroupware.org/bugzilla/show_bug.cgi?id=928

I think this is so easy that we can do it for OGo 1.0. But we need 
someone to try the implementation (or even better: find someone who can 
install an Asterisk reference server for developer testing on an OGo 
machine).

> The second is to use the "Manager API", which looks like a part of the 
> standard Asterisk server which (when enabled) listens on a TCP port 
> for commands to do stuff.  It's a simple text based protocol, which 
> includes status information about the call.  
> http://www.voip-info.org/wiki-Asterisk+manager+dialout

Should be easy as well, but here we definitely need a server for 
testing that. Captured at:
   http://bugzilla.opengroupware.org/bugzilla/show_bug.cgi?id=929

> Your point (in another post) about having the ip address associated 
> somehow with the source of the call is right on... unless we are 
> talking about mobile phones, (including analog portable handsets)... 
> in that case, it should be associated with the user, NOT the ip 
> address.
> We should have room for both situations, perhaps as a default per user?

I guess so. One step after the other ;-)

> Should this discussion be moved to the developer list?

Yes, maybe. Results of further investigation should also be added as 
comments to the Bugzilla entries.

Greets,
   Helge
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