Re: [Asterisk-Dev] RFC2833 Event Duration
Ryan Courtnage wrote: ... to it. In other words, Asterisk isn't passing along the duration of the original event. Correct. Is this by design? Can * be made to pass on the initial event duration? It is by design, i.e. not accidental. There is no way to pass on the duration at this time, because incoming events are turned into AST_FRAME_DTMF as they pass through the Asterisk core, and those frames do not currently have any way to carry duration information. Keep in mind that Asterisk is used in many applications that are not SIP-SIP, so just because in your situation there is an 'incoming event duration' that doesn't mean there is in all other applications :-) .. where we are unable to pass variable duration DTMF events to the switch, because Asterisk does not send along the duration. Our primary issue is that the duration is simply too short. There are a great many (most) PBXes that don't support variable DTMF event duration at all, so Asterisk is no different in this regard. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] RFC2833 Event Duration
On Saturday 05 November 2005 14:17, Kevin P. Fleming wrote: It is by design, i.e. not accidental. There is no way to pass on the duration at this time, because incoming events are turned into AST_FRAME_DTMF as they pass through the Asterisk core, and those frames do not currently have any way to carry duration information. Keep in mind that Asterisk is used in many applications that are not SIP-SIP, so just because in your situation there is an 'incoming event duration' that doesn't mean there is in all other applications :-) Why not just specify a reasonable duration? Or put it in sip.conf on a global/peer basis? -A. ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] RFC2833 Event Duration
Hello all, When holding down a digit on my (rfc2833) SIP phone, I see that (using Ethereal) the phone will send: RFC2833 RTP Event EventID: DTMF Five 5 (5) -- Duration: 47040 -- ... to Asterisk, where 47040 matches the amount of time I held down the 5 digit. If I've called another SIP device via Asterisk, Asterisk will only send: RFC2833 RTP Event EventID: DTMF Five 5 (5) -- Event Duration: 800 -- ... to it. In other words, Asterisk isn't passing along the duration of the original event. Is this by design? Can * be made to pass on the initial event duration? The duration can be manually adjusted in rtp.c's ast_rtp_senddigit function: /* Make duration 800 (100ms) */ rtpheader[3] |= htonl((800)); Is it safe to increase this value to say 4000? Any potential side-effects? If you are curious, we see an issue with: SIP Phone -- * -- SIP-PSTN switch .. where we are unable to pass variable duration DTMF events to the switch, because Asterisk does not send along the duration. Our primary issue is that the duration is simply too short. TIA Ryan ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
