[Freeswitch-users] How to interface with google talk

[Michael Jerris]

> From: danish.samad at vocalseeds.com [mailto:danish.samad at vocalseeds.com]
> 
> Mike, thanks for showing your concern. I was able to successfully
bridge a
> call between a google talk client and asterisk(SIP). Although there
were
> times when asterisk received and accepted the call while the jingle
> session was dropped. I did not really delve into the issue, so I am
not
> really sure what was causing the issue.
> 

I would like to see some traces of what is going on in freeswitch's
debug output when this happens.


> There are certain things I noticed, that I would like to know more
about.
> Firstly, when calling from jinlge to sip,I can only map a single
jabber
> account with a SIP account. Is there any way I can bridge multiple
calls
> between Jingle and SIP endpoints, using the current configuration. If
not,
> are you guys planning to implement this feature.
> 

The main issue here is a limitation in the jingle protocol, and the
gtalk client.  If using mod_dingaling on both sides with freeswitch, we
have an extension to the protocol that lets you pass the extension.  If
using gtalk, you would need a way to gather digits such as in a
javascript and launch that call to an extension.  The dingaling channel
will take text messages in the call beginning with + to be dtmf.

> Secondly, how can I make calls from sip endpoints to Jingle endpoints.
Any
> configuration tips will be helpful.
> 

You need to do it in some way that you know what destination and profile
will be used to call out, as for dingaling you need to specify the
sending profile.  You could just hardcode it to a specific profile.  We
use regex to parse a string like profilename!username!domain.com and
then send that as sip, parse the 3 parts using regex, and send it to the
bridge command for dingaling.

Mike.