I am using the audiocodes MP-108 FXO as a SIP gateway to the pstn, it works fine, and asterisk can receive and make calls thru it, caller id is also working ok. As for now the only remaining issue that i have pending is with conferences, the channels of the lines that get into a conference doesn't get freed in the gateway nor asterisk after hangin up, i'm going to try a 'exten => confexten,somepriority,Hangup()' after the Meetme command. This seems to be an issue with SIP, because i don't have that problem with my X100P FXO card.
John Sellens wrote:
I'm proposing an asterisk configuration and considering the use of multiport SIP/FXS gateways (instead of T1 cards and channel banks). I'm looking for products similar in function to the Cisco ATA-186, but with more ports.
I've seen the manufacturer's web pages for the Audiocodes MediaPack (http://www.audiocodes.com/) and the Mediatrix (http://www.mediatrix.com/) access devices.
Does anyone have any experiences to share with these or similar devices, or opinions on the relative merits of gateways vs channel banks?
Thanks very much!
John [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
