From: danish.samad@vocalseeds.com [mailto:danish.samad@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.