Re: [Asterisk-Users] nested dial, or jump to another line to continue dialing.
Joseph wrote: Is it possible to do nested dial() command on one line, Dial number, wait new seconds, dial another number etc. or dial number and jump to another line to continue dialing. D(ww) doesn't work as it sends DTMF but before the call is bridged, and I need to send numbers after the call is bridged. If you do a show application dial at the CLI: snip 'D([digits])' -- Send DTMF digit string *after* called party has answered but before the bridge. (w=500ms sec pause) Hmm.. it does say that DTMF is sent *after* called party has answered. it's been working for me since asterisk-1.0-RC2 Flynn ___ --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
Re: [Asterisk-Users] nested dial, or jump to another line to continue dialing.
On Wed, 2005-08-31 at 21:05 +0800, El Flynn wrote: Joseph wrote: Is it possible to do nested dial() command on one line, Dial number, wait new seconds, dial another number etc. or dial number and jump to another line to continue dialing. D(ww) doesn't work as it sends DTMF but before the call is bridged, and I need to send numbers after the call is bridged. If you do a show application dial at the CLI: snip 'D([digits])' -- Send DTMF digit string *after* called party has answered but before the bridge. (w=500ms sec pause) Hmm.. it does say that DTMF is sent *after* called party has answered. it's been working for me since asterisk-1.0-RC2 Flynn Yes, it dial the DTMF after the phone is answered. But I'm suing Sipura-3000 So what happens is, the calls come in, the sipura PSTN line answer the call and forwards it without delay to asterisk context. The problem is that the DTMF is dialed before the call is forwarded / bridged with asterisk extension. Doesn't matter what delay I insert. If the delay is long the message is being played not from the beginning. so I don't understand why it doesn't work. I've even try to accomplish this with Macro, it doesn't work. -- #Joseph ___ --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
Re: [Asterisk-Users] nested dial, or jump to another line to continue dialing.
On Wed, 2005-08-31 at 21:05 +0800, El Flynn wrote: Joseph wrote: Is it possible to do nested dial() command on one line, Dial number, wait new seconds, dial another number etc. or dial number and jump to another line to continue dialing. D(ww) doesn't work as it sends DTMF but before the call is bridged, and I need to send numbers after the call is bridged. If you do a show application dial at the CLI: snip 'D([digits])' -- Send DTMF digit string *after* called party has answered but before the bridge. (w=500ms sec pause) Hmm.. it does say that DTMF is sent *after* called party has answered. it's been working for me since asterisk-1.0-RC2 Flynn Here is a session with D() exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww218)) Executing Dial(SIP/11-3dec, SIP/[EMAIL PROTECTED]|30|D(ww218)) in new stack -- Called [EMAIL PROTECTED] -- SIP/pstn-5665-713c is ringing -- SIP/pstn-5665-713c answered SIP/11-3dec -- Goto (office-open,s,1) -- Executing Wait(SIP/pstn-1270-e0f5, 2) in new stack -- Attempting native bridge of SIP/11-3dec and SIP/pstn-5665-713c -- Executing Answer(SIP/pstn-1270-e0f5, ) in new stack -- Executing NVBackgroundDetect(SIP/pstn-1270-e0f5, welcome|t) in new stack -- Playing 'welcome' (language 'en') -- Executing Goto(SIP/pstn-1270-e0f5, 1|1) in new stack -- Goto (office-open,1,1) It is not passing DTMF(218) ---end session D()- Here is Macro session: exten = _51,1,Dial(SIP/[EMAIL PROTECTED],,TM(continue)) [macro-continue]; exten = s,1,Wait(5) exten = s,2,SendDTMF(218) Executing Dial(SIP/11-fe6c, SIP/[EMAIL PROTECTED]||TM(continue)) in new stack -- Called [EMAIL PROTECTED] -- SIP/pstn-5665-c577 is ringing -- SIP/pstn-5665-c577 answered SIP/11-fe6c -- Executing Wait(SIP/pstn-5665-c577, 5) in new stack -- Goto (office-open,s,1) -- Executing Wait(SIP/pstn-1270-c291, 2) in new stack -- Executing SendDTMF(SIP/pstn-5665-c577, 218) in new stack -- Executing Answer(SIP/pstn-1270-c291, ) in new stack -- Executing NVBackgroundDetect(SIP/pstn-1270-c291, welcome|t) in new stack -- Playing 'welcome' (language 'en') -- Attempting native bridge of SIP/11-fe6c and SIP/pstn-5665-c577 -- end Macro session-- Here is the what should happen, Manual Dial ext. 218 after connecting. Called [EMAIL PROTECTED] -- SIP/pstn-5665-d4cd is ringing -- SIP/pstn-5665-d4cd answered SIP/11-5068 -- Attempting native bridge of SIP/11-5068 and SIP/pstn-5665-d4cd -- Goto (office-open,s,1) -- Executing Wait(SIP/pstn-1270-acca, 2) in new stack -- Executing Answer(SIP/pstn-1270-acca, ) in new stack -- Executing NVBackgroundDetect(SIP/pstn-1270-acca, welcome|t) in new stack -- Playing 'sys_concept_welcome' (language 'en') Aug 31 15:56:31 NOTICE[22090]: app_nv_backgrounddetect.c:242 nv_background_detect_exec: DTMF received (matching to exten) -- #Joseph ___ --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
Re: [Asterisk-Users] nested dial, or jump to another line to continue dialing.
Joseph wrote: Here is a session with D() exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww218)) Executing Dial(SIP/11-3dec, SIP/[EMAIL PROTECTED]|30|D(ww218)) in new stack -- Called [EMAIL PROTECTED] -- SIP/pstn-5665-713c is ringing -- SIP/pstn-5665-713c answered SIP/11-3dec -- Goto (office-open,s,1) -- Executing Wait(SIP/pstn-1270-e0f5, 2) in new stack -- Attempting native bridge of SIP/11-3dec and SIP/pstn-5665-713c -- Executing Answer(SIP/pstn-1270-e0f5, ) in new stack -- Executing NVBackgroundDetect(SIP/pstn-1270-e0f5, welcome|t) in new stack -- Playing 'welcome' (language 'en') -- Executing Goto(SIP/pstn-1270-e0f5, 1|1) in new stack -- Goto (office-open,1,1) It is not passing DTMF(218) ---end session D()- Without looking at your dialplan for the context that SIP/4791270 belongs to, what's most likely happening is the pauses are too short or too long. I've just dialed out to my cellphone to test this, and yes i do hear the DTMF when I pick up my cellphone. You can create a simple test for this -- dial to your mobile phone or landline, pick up the call and see if the DTMF is passed. Flynn ___ --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
Re: [Asterisk-Users] nested dial, or jump to another line to continue dialing.
On Thu, 2005-09-01 at 10:07 +0800, El Flynn wrote: Joseph wrote: Here is a session with D() exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww218)) Executing Dial(SIP/11-3dec, SIP/[EMAIL PROTECTED]|30|D(ww218)) in new stack -- Called [EMAIL PROTECTED] -- SIP/pstn-5665-713c is ringing -- SIP/pstn-5665-713c answered SIP/11-3dec -- Goto (office-open,s,1) -- Executing Wait(SIP/pstn-1270-e0f5, 2) in new stack -- Attempting native bridge of SIP/11-3dec and SIP/pstn-5665-713c -- Executing Answer(SIP/pstn-1270-e0f5, ) in new stack -- Executing NVBackgroundDetect(SIP/pstn-1270-e0f5, welcome|t) in new stack -- Playing 'welcome' (language 'en') -- Executing Goto(SIP/pstn-1270-e0f5, 1|1) in new stack -- Goto (office-open,1,1) It is not passing DTMF(218) ---end session D()- Without looking at your dialplan for the context that SIP/4791270 belongs to, what's most likely happening is the pauses are too short or too long. I've just dialed out to my cellphone to test this, and yes i do hear the DTMF when I pick up my cellphone. You can create a simple test for this -- dial to your mobile phone or landline, pick up the call and see if the DTMF is passed. Flynn Here is my dial plan, just for testing. I have two land lines: exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww218)) Though when I was looking around I run onto another user experiencing the same, as I have: http://voxilla.com/index.php?name=PNphpBB2file=viewtopict=3515highlight=rfc2833+dtmf ---quote I recently noticed that my Sipura SPA-3000 refused to play DTMF keys sent to it by Asterisk (e.g. using, in extensions.conf, the SendDTMF() application or the D() parameter in the Dial() command). This applies to all versions of Asterisk, at least until the CVS HEAD of the 11th of May, 2005. After a lot of testing, I discovered that the SPA-3000 doesn't like to receive event start and event end RTP Event packets with the same timestamp. In versions up to the stable release 1.0.7, the timestamp field was not updated at all by the routines sending DTMF digits, and that represented a separate bug fixed in the CVS about thre weeks ago (see http://bugs.digium.com/view.php?id=3675 ). However, the ast_rtp_senddigit() function in rtp.c still uses the same timestamp for all six RTP packets (three copies of event start and three copies of event end) sent to the receiving peer. When it receives those packets, the SPA-3000 fails to produce a DTMF tone; instead, it produces a sort of click for each digit. In order to work around this problem, the timestamp field of the three last packets must be at least one unit higher than the one in the first three packets. Accordind to the section 3.5 of RFC2833, both Asterisk and SPA-3000 get it wrong: the correct semantics should be that each packet instructs the receiver to play a tone from the time written in the timestamp to that time plus the content of the duration field; successive packets may refine the two limits but not beyond the end determined by timestamp plus duration in the event end packet(s). A separate minor problem is that the SPA-3000 appears to ignore both the duration calculated from the RTP event packets and the content of the setting DTMF Playback Length: in the Regional screen of the advanced admin setup, and always plays a tone about 500 ms long. In order to leave a gap between the digits, one should pass to asterisk strings with the digits separated by 'w' characters (which translate into 500 ms pauses). For instance: Code: exten = **802,2,SendDTMF(1w6w2w5w#) BTW, I'm curious to know if anybody else has experienced such problems. -end quote- Though, this solution didn't work for me either, inserting w between digits. -- #Joseph ___ --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] nested dial, or jump to another line to continue dialing.
Is it possible to do nested dial() command on one line, Dial number, wait new seconds, dial another number etc. or dial number and jump to another line to continue dialing. D(ww) doesn't work as it sends DTMF but before the call is bridged, and I need to send numbers after the call is bridged. -- #Joseph ___ --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
