If you put canreinvite=no in the sip peer entry for your FXO gateway, you would ensure the voice stream goes through Asterisk.
> -----Original Message----- > From: Syd Carter [mailto:[EMAIL PROTECTED] > Sent: April 19, 2007 9:02 PM > To: [email protected] > Subject: [on-asterisk] Audio, Endpoints & Octasic soft Echo solution > > Hi all. Just need a bit of advice regarding the routing of a > call. You see, I've installed the Octasic soft echo > solution. Unfortunately it only works with zap channels. My > setup only does not have zap channels because I connect using > external ATA devices such as the SPA3000. > Anyways, I've been in contact with the tech support at > Octasic about this and they are interested in resolving echo > for setups such as mine, but need a bit of advice regarding > the audio path between endpoints. > > Can anyone provide insight into their question: > /"Can you confirm if the voice streams go through the > asterisk box. Our understanding is that once the SIP call is > set up, the endpoints will be exchanging voice streams > directly. Any insight you may provide will be welcome." > > /My opinion is that the voice stream is going through the > box, because 1. I have enabled "T" in my dial command. > 2. I've enabled for recording of calls. > > However I'm thinking that they need to know something more > concrete that if the voice streams go through the asterisk > box. Anyone care to add their 2 cents worth that I could > send back in a reply? Thanks.. Syd > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] For > additional commands, e-mail: [EMAIL PROTECTED] >
