Re: [Asterisk-Dev] monitoring a call and media path
Denis Smirnov wrote: > With last changes to IAX2, can this be fixed for SIP? It can be fixed for SIP, but the changes for IAX2 are irrelevant. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] monitoring a call and media path
On Mon, Dec 05, 2005 at 07:37:55PM -0600, Kevin P. Fleming wrote: >>I was thinking of hacking things a bit to allow my asterisk to stay >>out of the media path in the above case, but figured it couldn't hurt >>to post a quick sanity check here. Anyone see any problems? KPF> This is certainly possible, but Asterisk currently assumes that if it is KPF> not in the media path, it also won't be able to receive DTMF frames. KPF> However, if you are using SIP INFO for DTMF signaling, then it should KPF> 'just work', since when Asterisk sees the appropriate DTMF frames it KPF> will cause the bridge to 'break' and bring the media path back. With last changes to IAX2, can this be fixed for SIP? -- JID: [EMAIL PROTECTED] ICQ: 58417635 (please, use jabber, if you can) http://freesource.info/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] monitoring a call and media path
Wolfgang S. Rupprecht wrote: I was thinking of hacking things a bit to allow my asterisk to stay out of the media path in the above case, but figured it couldn't hurt to post a quick sanity check here. Anyone see any problems? This is certainly possible, but Asterisk currently assumes that if it is not in the media path, it also won't be able to receive DTMF frames. However, if you are using SIP INFO for DTMF signaling, then it should 'just work', since when Asterisk sees the appropriate DTMF frames it will cause the bridge to 'break' and bring the media path back. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] monitoring a call and media path
On a purely SIP call between two sip phones with canreinvite=yes and the dialing on both phones being done via SIP info commands, is there any need to keep asterisk in the RTP media path if no recording is being made but one wants to allow a later recording via *1? Currently asterisk seems to stay in the loop so it can catch inband dialing, but that isn't always needed if the phones cooperate and put the dialing into sip msgs. I was thinking of hacking things a bit to allow my asterisk to stay out of the media path in the above case, but figured it couldn't hurt to post a quick sanity check here. Anyone see any problems? -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
