I (and some others) are having an issue with placing calls on hold.
Our setup is as follows: IAX2 or SIP terminator/originator ---> asterisk box ---> SIP Phones I have tried asterisk verison 1.0.9, 1.2.6 and 1.2.12, and all have the same issue. When I place a call on hold that has come in a PSTN channel (through a PRI and a Digium card) everything is fine. When I place a call on hold that has come in or gone out an IAX2 or SIP terminator or originator, when I pick the call back up often there is one-way-audio (I can hear the caller, but the caller can not hear me). I have attached a packet capture of the situation, and as you can see at about packet #3803 the audio goes one way. Can anyone enlighten me as to why this is happening, and why asterisk is no longer sending audio back to the terminator? I would like to get this fixed, obviously. (This file was a tcpdump, and can be opened in ethereal/wireshark) EDIT: Packet capture at following address: http://hecate.chilitech.net/~matth/zootdump.cap _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
