Christian,
Thanks for your response.
In my case, I was asked to do it through SIP phone 3 way call functionality
and not the Asterisk Meetme application.
I wanted to know if any one had done something similar in past or not.
I am short of PRI in my test environment and hence I can't test it
Will someone help/direct me find a way to implement this?
Or you can suggest some other method.
On Fri, Dec 17, 2010 at 12:44 PM, Asterisk Man theasterisk...@gmail.comwrote:
Hi friends,
I want to implement following scenario using Asterisk. Please suggest me
whether it is possible or
not.
Without reading too much into your description, I can tell you that being an
inband sound, and as long as the dtmf tone is heard by everybody during the
conference, and being the ivr gateway one of the parties of the conference, I
don't see a reason why the ivr gateway wouldn't act upon hearing
Hi friends,
I want to implement following scenario using Asterisk. Please suggest me
whether it is possible or
not.
This is bit off Asterisk and more on SIP side.
An Asterisk box with one Station(SIP channel) and PRI.
Agent dials a PSTN number of customer from station through Asterisk PRI.
