Figured out the Packet2Packet mystery. That log message is no longer in the 1.8 source code, so you will never see it. You can however see that it is working as "Sent RTP P2P packet" if you do an "rtp set debug on" (not a good idea on a busy system).
I sat down with Nabeel this morning, and he pointed out that I have "T" option to my Dial() command, which I'm using for shortcuts defined in features.conf. Unfortunately asterisk inspects all the RTP packets, not just the RFC2833 ones when the T option is on -- and so I wasn't seeing either P2P or Packet2Packet log messages. Any other DTMF pointers beyond removing the "T" option? On 1 May 2011 22:17, Simon P. Ditner <[email protected]> wrote: > Maybe I am doing something wrong. > > I don't see either a "Packet2Packet bridging" or "Native bridging" on > the console when the call comes up. I have canreinvite=no, so I don't > expect to see native bridging. Verbosity is set to 5. > > Packet captures are showing everything using g711 ulaw, 20ms packet > sizes on all legs. > > Any debugging tips, or extra options to toggle in sip.conf or rtp.conf > to get it to try Packet2Packet? > > On 1 May 2011 20:44, Matthew Gamble <[email protected]> wrote: >> Packet2Packet Bridging is when audio only goes into the RTP stack and >> directly out, bypassing the Asterisk core. Without it, Asterisk will >> use native bridging and will fully process the audio between the two >> endpoints. >> >> For Packet2Packet bridging to work, both sides of the call must be >> using the same channel type, codec, and packet size, otherwise native >> bridging is required to convert the audio between the two sides. >> >> >> >> On Sun, May 1, 2011 at 8:33 PM, Simon P. Ditner <[email protected]> wrote: >>> This is the first I've heard of "packet to packet bridging", and so >>> far haven't found an explanation of how it's controlled, or when it's >>> present or disabled. Is this a direct config option, or a side effect >>> of some other option? >>> >>> On 28 April 2011 22:28, <[email protected]> wrote: >>>> And what DTMF mode are you trying to use on both the SIP Trunk and the >>>> device? With RFC2833 on both side and Packet to Packet bridging you >>>> shouldn't have any issues. >>>> >>>> >>>> ------Original Message------ >>>> From: [email protected] >>>> To: Simon P. Ditner >>>> To: [email protected] >>>> To: 'Asterisk Users Group' >>>> ReplyTo: [email protected] >>>> Subject: Re: [on-asterisk] DTMF, will it ever work? >>>> Sent: Apr 28, 2011 22:27 >>>> >>>> Which version of asterisk are you using? >>>> ------Original Message------ >>>> From: Simon P. Ditner >>>> Sender: [email protected] >>>> To: 'Asterisk Users Group' >>>> Subject: [on-asterisk] DTMF, will it ever work? >>>> Sent: 28 Apr 2011 22:22 >>>> >>>> Is there some trick to getting asterisk working reliably when calling >>>> into conference bridges and voicemail? No end of trouble with DTMF on >>>> my home system regardless of which provider I'm using. Drives my wife >>>> nuts when she's working from home. >>>> >>>> --------------------------------------------------------------------- >>>> To unsubscribe, e-mail: [email protected] >>>> For additional commands, e-mail: [email protected] >>>> >>>> >>>> >>>> >>>> --------------------------------------------------------------------- >>>> To unsubscribe, e-mail: [email protected] >>>> For additional commands, e-mail: [email protected] >>>> >>>> >>>> Sent from my BlackBerry device on the Rogers Wireless Network >>> >> >> --------------------------------------------------------------------- >> To unsubscribe, e-mail: [email protected] >> For additional commands, e-mail: [email protected] >> >> > --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
