search for: playdtmf

Hi sorry to bother, but I wasted a lot of time on this question, contact several forum (as much english as french), and still no answer :-(. In order to simulate an xfert, I thought that the PlayDTMF manager command will be the right one. All seems to be perfect : - I have the sound associated to the simulated key - I have the message : DTMF successfully queued But, nothing happen (no xfert, no voice). Did somebody use this command before, or have an idea ? P.S. : - asterisk 1.4.6 (make, mak...

Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1 the result is: <ajax-response> <response type='object' id='unknown'><generic response='Success' message=...

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer *2 One Touch Monitor...

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer *2 One Touch Monitor...

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/595457d0/attachment.htm

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer *2 One Touch Monitor...

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer *2 One Touch Monitor...

...otify Crash on 16.6 (Reported by Joshua Elson) * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) * ASTERISK-28561 - Asterisk Deadlocks (Reported by Aheliotech) * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell) * ASTERISK-28566 - CDR backend unload problem during active call(s) (Reported by Marian Piater) * ASTERISK-28544...

...wi container (Reported by Kevin Harwell) * ASTERISK-28566 - CDR backend unload problem during active call(s) (Reported by Marian Piater) * ASTERISK-28553 - stasis.c: Crash during unload (Reported by Kevin Harwell) * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) * ASTERISK-28544 - Wrong contact representation in ipv6 mode (Reported by J��rgen H) * ASTERISK-28534 - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden) * ASTERISK-28463 - res_pj...

...otify Crash on 16.6 (Reported by Joshua Elson) * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) * ASTERISK-28561 - Asterisk Deadlocks (Reported by Aheliotech) * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd) * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell) * ASTERISK-28566 - CDR backend unload problem during active call(s) (Reported by Marian Piater) * ASTERISK-28553...

Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

...parameter / "maxredirs" doesn't do anything (Reported by candrews) * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-...

...parameter / "maxredirs" doesn't do anything (Reported by candrews) * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-...

...parameter / "maxredirs" doesn't do anything (Reported by candrews) * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl) * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec) * ASTERISK-28533 - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp) * ASTERISK-...

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

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