Re: [asterisk-users] SIP DTMF Flash Event
Tim Nelson wrote: Is there a way to have Asterisk respond appropriately when receiving a DTMF Flash event via SIP? I'm finding some WiFi SIP phones, specifically the Quickphones QA-342 want to handle transfers/3-way calls by sending a DTMF Flash event instead of handling it properly like every other damn VoIP phone on the planet... Asterisk sees the Flash event (via the logs), but does not act upon it. Thoughts? Hola, The functionality you are talking about (server side SIP transfers and conferences) are not really implemented for use like this. All of the pieces exist within Asterisk to achieve the expected end result but as this has only ever come up maybe twice noone has ever taken the time to do it. Sorry! Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP DTMF Flash Event
Is there a way to have Asterisk respond appropriately when receiving a DTMF Flash event via SIP? I'm finding some WiFi SIP phones, specifically the Quickphones QA-342 want to handle transfers/3-way calls by sending a DTMF Flash event instead of handling it properly like every other damn VoIP phone on the planet... Asterisk sees the Flash event (via the logs), but does not act upon it. Thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF problem with SNOM (solved)
Hi! After heavy debugging I found out that I had packet loss from the SNOM phone to Asterisk. Connecting the SNOM phone from the office switch (with autodetect) to an HP Procurve (with autodetect) solved the issue - packet loss is gone and DTMF works. regards klaus Klaus Darilion schrieb: Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042766, ts 4066332328, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042774, ts 4066333608, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042775, ts 4066333768, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042777, ts 4066334088, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042778, ts 4066334248, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00320) Got RTP packet from83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01120) Got RTP packet from83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01280) Got RTP packet from83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01600) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 00, seq 042790, ts 4066336648, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042791, ts 4066336808, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042792, ts 4066336968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042793, ts 4066337128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042794, ts 4066337288, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042795, ts 4066337448, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042798, ts 4066337928, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042801, ts 4066338408, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042802, ts 4066338568, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042805, ts 4066339048, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042806, ts 4066339208, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042809, ts 4066339688, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04) Got RTP RFC2833 from
[asterisk-users] SIP DTMF problem with SNOM
Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042766, ts 4066332328, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042774, ts 4066333608, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042775, ts 4066333768, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042777, ts 4066334088, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042778, ts 4066334248, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00320) Got RTP packet from83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01120) Got RTP packet from83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01280) Got RTP packet from83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01600) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 00, seq 042790, ts 4066336648, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042791, ts 4066336808, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042792, ts 4066336968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042793, ts 4066337128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042794, ts 4066337288, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042795, ts 4066337448, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042798, ts 4066337928, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042801, ts 4066338408, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042802, ts 4066338568, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042805, ts 4066339048, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042806, ts 4066339208, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042809, ts 4066339688, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04, mark 0, event 0002, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042813, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042813, ts 4066340168, len 04, mark 0, event 0002, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq
Re: [asterisk-users] SIP DTMF problem with SNOM
How are you testing DTMF detection with the Snom UA? Klaus Darilion wrote: Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042766, ts 4066332328, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042774, ts 4066333608, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042775, ts 4066333768, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042777, ts 4066334088, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042778, ts 4066334248, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00320) Got RTP packet from83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01120) Got RTP packet from83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01280) Got RTP packet from83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01600) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 00, seq 042790, ts 4066336648, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042791, ts 4066336808, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042792, ts 4066336968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042793, ts 4066337128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042794, ts 4066337288, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042795, ts 4066337448, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042798, ts 4066337928, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042801, ts 4066338408, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042802, ts 4066338568, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042805, ts 4066339048, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042806, ts 4066339208, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042809, ts 4066339688, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04, mark 0, event 0002, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042813, ts 4066340168, len 04) Got RTP RFC2833
Re: [asterisk-users] SIP DTMF problem with SNOM
I have a similar problem with Snom. Since I've upgraded from version 6 to version 7 I cannot call IVR systems. The first DTMF goes ok, but after that others are not accepted nor I am heard by the operator at the other side. Since I am the only one who has Snom here I didn't bother to debug it... __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF problem with SNOM
I have a Snom 320 and both inband and RFC2833 OOB work fine for me. Yehavi Bourvine wrote: I have a similar problem with Snom. Since I've upgraded from version 6 to version 7 I cannot call IVR systems. The first DTMF goes ok, but after that others are not accepted nor I am heard by the operator at the other side. Since I am the only one who has Snom here I didn't bother to debug it... __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF problem with SNOM
Alex Balashov schrieb: How are you testing DTMF detection with the Snom UA? The Voicemail(u...@context) application asks the user for the voicemail password. Using eyebeam everyhing works fine. Using SNOM (320 FW 7.3.10a) it works almost never. regards klaus Klaus Darilion wrote: Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042766, ts 4066332328, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042774, ts 4066333608, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042775, ts 4066333768, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042777, ts 4066334088, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042778, ts 4066334248, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042780, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00320) Got RTP packet from83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042781, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00480) Got RTP packet from83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042782, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 00640) Got RTP packet from83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042785, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01120) Got RTP packet from83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042786, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01280) Got RTP packet from83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042788, ts 4066334648, len 04, mark 0, event 0001, end 0, duration 01600) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04) Got RTP RFC2833 from 83.136.33.3:64118 (type 101, seq 042789, ts 4066334648, len 04, mark 0, event 0001, end 1, duration 01760) Got RTP packet from83.136.33.3:64118 (type 00, seq 042790, ts 4066336648, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042791, ts 4066336808, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042792, ts 4066336968, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042793, ts 4066337128, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042794, ts 4066337288, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042795, ts 4066337448, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042798, ts 4066337928, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042801, ts 4066338408, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042802, ts 4066338568, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042805, ts 4066339048, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042806, ts 4066339208, len 000160) Got RTP packet from83.136.33.3:64118 (type 00, seq 042809, ts 4066339688, len 000160) Got RTP packet from83.136.33.3:64118 (type 101, seq 042812, ts 4066340168, len 04) Got RTP RFC2833 from 83.136.33.3:64118
[asterisk-users] SIP # DTMF
Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only 333 What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft für Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gestión Pública Descentralizada y Lucha Contra La Pobreza - PADEP Av. Sánchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email: [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP # DTMF
core show application dial (this is the official application doc) Pay special attention to the D() option. Rodolfo Alcazar Portillo wrote: Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only 333 What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Consulting and design services for LAN, WAN, voice and data. Based near Birmingham, AL. Now accepting clients worldwide. Contact me for Tellabs echo canceling systems. Also see http://www.fnords.org/skillslist.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP # DTMF
On many phones # sends the call. Rodolfo Alcazar Portillo wrote: Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only 333 What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Thank you and have any kind of day you want, Anthony Francis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP DTMF Troubleshoot
Everything seems find on my end. Here's the setup:
Linksys SPA922 - Asterisk 1.4 --- Quintum T1 gateway
Between Asterisk and Quintum if I use G729 RFC2833 DTMF works with no
issues, however if I use uLaw this is where there is a problem. For
some reason the Quintum gateway does not support uLaw + RFC2833.
Also does not matter if I use Asterisk 1.2 or a grandstream or the
proverbial SIP tin can; The scenario is always the same.
On Jan 28, 2008 7:03 PM, Alex Balashov [EMAIL PROTECTED] wrote:
I think your best bet is to do a packet capture and look for RTP packets
with an RTP Event payload (rtpevent display filter).
On Mon, 28 Jan 2008, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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[asterisk-users] SIP DTMF Troubleshoot
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
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Re: [asterisk-users] SIP DTMF Troubleshoot
On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
I'd first turn on rtp debug and see if that helps. If that's not
enough information, I'd go into logger.conf and add dtmf to the logger
and messages lines (and any others you care about), and then do a
logger reload from the Asterisk CLI.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] SIP DTMF Troubleshoot
Too much info then too little info.
Basically the issue is the provider this happens even when we send
them the calls in IAX because they talk SIP to the same gateway.
I just need to prove it to these people. Anyone have any DTMF issues
between Asterisk and a Quintum gateway?
On Jan 28, 2008 6:47 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
I'd first turn on rtp debug and see if that helps. If that's not
enough information, I'd go into logger.conf and add dtmf to the logger
and messages lines (and any others you care about), and then do a
logger reload from the Asterisk CLI.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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Re: [asterisk-users] SIP DTMF Troubleshoot
I think your best bet is to do a packet capture and look for RTP packets
with an RTP Event payload (rtpevent display filter).
On Mon, 28 Jan 2008, Andrew Joakimsen wrote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways but how to troubleshoot DTMF issues?
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Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671
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[asterisk-users] SIP DTMF not acted on for features in 1.4.0b3
Asterisk seems to be ignoring DTMF for features in Asterisk 1.4.0b3 My SNOM sends the dtmf-relay and Asterisk gets it because I can see it in the sip debug. However, once seen, Asterisk doesn't actually do anything about it. I've checked features and that seems fine. Is this a bug or something that I've screwed up? For the record, here's the features setting: asterisk*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# # Attended Transfer *2 One Touch Monitor *1 Disconnect Call * * Park Call #72 Dynamic Feature Default Current --- --- --- testfeature no def #9 Call parking Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-720 asterisk*CLI and here's a SIP trace of me pressing '*' during a call (which according to my features should Disconnect the Call. asterisk*CLI --- SIP read from 192.168.1.12:5060 --- INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;rport From: Russell 112 sip:[EMAIL PROTECTED];tag=tmyszljbna To: sip:[EMAIL PROTECTED];tag=as0b7389e4 Call-ID: [EMAIL PROTECTED] CSeq: 14 INFO Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;line=gv8x1x75;flow-id=1 User-Agent: snom360/6.5.1 Content-Type: application/dtmf-relay Content-Length: 22 Signal=* Duration=160 - --- (11 headers 2 lines) --- Receiving INFO! * DTMF-relay event received: * asterisk*CLI --- Transmitting (no NAT) to 192.168.1.12:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-llnm8m4u2wef;received=192.168.1.12;rport=5060 From: Russell 112 sip:[EMAIL PROTECTED];tag=tmyszljbna To: sip:[EMAIL PROTECTED];tag=as0b7389e4 Call-ID: [EMAIL PROTECTED] CSeq: 14 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/112-0070a2c0] asterisk*CLI Can anyone suggest what's wrong here? Thanks. -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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
[Asterisk-Users] SIP DTMF problem
Hi! I have this setup: Analog phone - Audiocodes MP-114 - Asterisk 1- Aastra 480i | \/ Asterisk 2 The codec is alaw on all the calls. Asterisk is CVS-HEAD checked it a couple of hours ago. Asterisk 1 and 2 is connected with a SIP connection using INFO. The Aastra 480i does not support DTMF INFO mode as far as I can tell, so I configures it as rfc2833 in sip realtime. The MP-124 is confugured to use INFO. When I call the voicemail on Asterisk 2 from the analog phone it works fine. When I call from the 480i it does not work. Asterisk 1 and 2 does not hear the DTMF. When I call from the analog phone to the 480i or from the 480i to the analog phone Asterisk 1 hears the DTMF (* DTMF Received: '1') that is pressed on the analog phone and only some of the tones pressed on the 480i. Some of the DTMF sent from the 480i is registered multiple times on Asterisk 1 e.g. when I press 1 Asterisk register the 1 but thinks it is pressed 2 or 3 times, but I only pressed it once. Is there a problem when Asterisk needs to convert between INFO and rfc2833? Or am I missing something.-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP DTMF problem
Hi! I have this setup: Analog phone - Audiocodes MP-114 - Asterisk 1- Aastra 480i | \/ Asterisk 2 The codec is alaw on all the calls. Asterisk is CVS-HEAD checked it a couple of hours ago. Asterisk 1 and 2 is connected with a SIP connection using INFO. The Aastra 480i does not support DTMF INFO mode as far as I can tell, so I configures it as rfc2833 in sip realtime. The MP-124 is confugured to use INFO. When I call the voicemail on Asterisk 2 from the analog phone it works fine. When I call from the 480i it does not work. Asterisk 1 and 2 does not hear the DTMF. When I call from the analog phone to the 480i or from the 480i to the analog phone Asterisk 1 hears the DTMF (* DTMF Received: '1') that is pressed on the analog phone and only some of the tones pressed on the 480i. Some of the DTMF sent from the 480i is registered multiple times on Asterisk 1 e.g. when I press 1 Asterisk register the 1 but thinks it is pressed 2 or 3 times, but I only pressed it once. Is there a problem when Asterisk needs to convert between INFO and rfc2833? Or am I missing something.-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP dtmf=rfc2833 not working
We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that the IAXy is digitizing DTMF tones and sending just the pure data, rather than the audio representation, and that this explains why the IAXY's work flawlessly in this application. I am also under the impression that SIP modems should also support a mode like this.. We have tried: dtmfmode=rfc2833 in sip.conf, and we have also tried turning on DTMF Tx: to AVT on the Sipura, but this does not affect reliability at all. So my question is: 1) Are we doing anything wrong, or is there something more we should be doing, to enable DTMF translation (ala rfc2833) in Asterisk and/or our SIP modems? 2) Is there any kind of debugging mode in Asterisk which we can turn on, which will show once and for all whether or not we really have successfully enabled rfc2833? We are using Asterisk 1.0.3, by the way. Thank you very much in advance! Brent ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-DTMF
Hi, Can anyone please comment on this? --- Asterisk . [EMAIL PROTECTED] wrote: Hi Alex, --- Alex Barnes [EMAIL PROTECTED] wrote: You should set the type of DTMF on a per SIP PEER basis (sip.conf). Then simply set the SJPhone peer to use dtmfmode=inband. I have used SJPhone without problems along side Snoms that use dtmfmode=rfc2833. Thanks for the response. I know this will work, if the UACs are registered with Asterisk. But none of the UACs that dial this number are registered with Asterisk. They just use the sip uri to dial to that number. ie, like this: sip:[EMAIL PROTECTED]:port. I was trying to make any sip client to reach this number and to the desired extension just by dialing using the sip uri. Hope that explains the problem. Any help appreciated. Thanks again, Girish HTH Alex -Original Message- I have mapped a number in the default context of my dialplan. When someone dials that number, it plays an IVR message and allows the caller to enter 4 digit extensions. If the extension is a valid one, the call wll be routed to that particular extension. 'INFO' is set as the dtmf mode. This works fine if i call from a SIP UAC which sends dtmf as INFO. But When i dial using SJPhone, call doesn't get routed, because SJPhone uses inband dtmf. So, my problem is only people who use UACs that send dtmf using the INFO method can reach the desired extension, where as people who use SJPhone cannot do this. Can i make Asterisk to receive both info and inband dtmf for the same number? Is this possible? If so, can anyone tell me how to do that? Thanks, Girish ___ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP-DTMF
Hi, I have mapped a number in the default context of my dialplan. When someone dials that number, it plays an IVR message and allows the caller to enter 4 digit extensions. If the extension is a valid one, the call wll be routed to that particular extension. 'INFO' is set as the dtmf mode. This works fine if i call from a SIP UAC which sends dtmf as INFO. But When i dial using SJPhone, call doesn't get routed, because SJPhone uses inband dtmf. So, my problem is only people who use UACs that send dtmf using the INFO method can reach the desired extension, where as people who use SJPhone cannot do this. Can i make Asterisk to receive both info and inband dtmf for the same number? Is this possible? If so, can anyone tell me how to do that? Thanks, Girish __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-DTMF
You should set the type of DTMF on a per SIP PEER basis (sip.conf). Then simply set the SJPhone peer to use dtmfmode=inband. I have used SJPhone without problems along side Snoms that use dtmfmode=rfc2833. HTH Alex -Original Message- From: Asterisk . [mailto:[EMAIL PROTECTED] Sent: 27 October 2004 13:49 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP-DTMF Hi, I have mapped a number in the default context of my dialplan. When someone dials that number, it plays an IVR message and allows the caller to enter 4 digit extensions. If the extension is a valid one, the call wll be routed to that particular extension. 'INFO' is set as the dtmf mode. This works fine if i call from a SIP UAC which sends dtmf as INFO. But When i dial using SJPhone, call doesn't get routed, because SJPhone uses inband dtmf. So, my problem is only people who use UACs that send dtmf using the INFO method can reach the desired extension, where as people who use SJPhone cannot do this. Can i make Asterisk to receive both info and inband dtmf for the same number? Is this possible? If so, can anyone tell me how to do that? Thanks, Girish __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-DTMF
Hi Alex, --- Alex Barnes [EMAIL PROTECTED] wrote: You should set the type of DTMF on a per SIP PEER basis (sip.conf). Then simply set the SJPhone peer to use dtmfmode=inband. I have used SJPhone without problems along side Snoms that use dtmfmode=rfc2833. Thanks for the response. I know this will work, if the UACs are registered with Asterisk. But none of the UACs that dial this number are registered with Asterisk. They just use the sip uri to dial to that number. ie, like this: sip:[EMAIL PROTECTED]:port. I was trying to make any sip client to reach this number and to the desired extension just by dialing using the sip uri. Hope that explains the problem. Any help appreciated. Thanks again, Girish HTH Alex -Original Message- I have mapped a number in the default context of my dialplan. When someone dials that number, it plays an IVR message and allows the caller to enter 4 digit extensions. If the extension is a valid one, the call wll be routed to that particular extension. 'INFO' is set as the dtmf mode. This works fine if i call from a SIP UAC which sends dtmf as INFO. But When i dial using SJPhone, call doesn't get routed, because SJPhone uses inband dtmf. So, my problem is only people who use UACs that send dtmf using the INFO method can reach the desired extension, where as people who use SJPhone cannot do this. Can i make Asterisk to receive both info and inband dtmf for the same number? Is this possible? If so, can anyone tell me how to do that? Thanks, Girish ___ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP - DTMF Payload type
I have a problem with my Welltech Wellgates. I can't call any extension which starts with or includes * or #. When dialing it responds fine but after some seconds I just get a busy tone and on the Asterisk console it says SIP/2.0 484 Address Incomplete. Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload type was set to 96 (default) on my Wellgate, Asterisk responded with NOTICE[606227]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 96 received I then tried to set it to 101 (found this value somewhere on the net) and verified that voice responds now worked, but I don't know if this is the correct type? Still I can't use * or # regards Mickey Binder ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP - DTMF Payload type
Don't know if it connects to the DTMF payload type. Yesterday I made som different tests and observed that if DTMF payload type was set to 96 (default) on my Wellgate, Asterisk responded with NOTICE[606227]: File rtp.c, Line 417 (ast_rtp_read): Unknown RTP codec 96 received Just wanted to note I just observed it doesn't send any number at all when using # or *. In the field Contact it writes: sip:@10.1.1.51 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP, DTMF, and AudioCodes Mediant 2k
Greetings... I'm working on getting an AudioCodes Mediant 2000 big box o' PRI's going with Asterisk, and am running into a problem with DTMF handling. The box is sending DTMF packets to Asterisk as INFO packets, and they are apparently being seen by Asterisk. However, the DTMF knowledge doesn't seem to actually do anything -- the VM system doesn't recognize the digits, and if I call a SIP phone from the outside world, the beeps don't make it through. Here's what the packets look like, according to sip debug: Sip read: INFO sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 208.34.86.37;branch=z9hG4bKacyRPrQQu From:sip:[EMAIL PROTECTED];tag=1c11546 To: sip:[EMAIL PROTECTED];tag=as37f2b147 Call-ID: [EMAIL PROTECTED] CSeq: 3100834 INFO Supported: 100rel,em Content-Type: application/sdp Content-Length: 35 p=Digit-Collection y=Digits d=7 9 headers, 3 lines Receiving DTMF! Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.34.86.37;branch=z9hG4bKacyRPrQQu From: sip:[EMAIL PROTECTED];tag=1c11546 To: sip:[EMAIL PROTECTED];tag=as37f2b147 Call-ID: [EMAIL PROTECTED] CSeq: 3100834 INFO User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 208.34.86.37:5060 Sip read: INFO sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 208.34.86.37;branch=z9hG4bKacJRCRkbi From:sip:[EMAIL PROTECTED];tag=1c11546 To: sip:[EMAIL PROTECTED];tag=as37f2b147 Call-ID: [EMAIL PROTECTED] CSeq: 3100835 INFO Supported: 100rel,em Content-Type: application/sdp Content-Length: 35 p=Digit-Collection y=Digits d=0 9 headers, 3 lines Receiving DTMF! Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 208.34.86.37;branch=z9hG4bKacJRCRkbi From: sip:[EMAIL PROTECTED];tag=1c11546 To: sip:[EMAIL PROTECTED];tag=as37f2b147 Call-ID: [EMAIL PROTECTED] CSeq: 3100835 INFO User-Agent: Asterisk PBX Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 208.34.86.37:5060 My sip.conf entry looks like the following: [rochny-audiocodes1] type=friend host=208.34.86.37 dtmfmode=info secret=[mumble] context=inbound I've tried various settings for dtmfmode, there and in the [general] section, to no avail. Any help or troubleshooting tips would be appreciated. Thanks! :-) -rt -- Ryan Tucker [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP DTMF settings
What are the dtmf= options in sip.conf. --Eric ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
