similar to: PlayDTMF and Asterisk Manager

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='DTMF successfully queued' /></response>

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

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:

Hi everybody !!! I'm a french student working on an Asterisk's projet . I think it's very easy for you : configure 2 CISCO 7910 and Asterisk in order to make call between both. (Scuse for my poor english ;) ) I have install Debian and Asterisk thanks to the CD Xorcom Rapid. I have set-up DHCP and TFTP servers for the IP phones. On the tftp, i have put the *.bin files for my phone,

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

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

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

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

In the capi.conf file, there is a bridge option that allow to "native bridging (CAPI line interconnect) if available", and I found this in the capi-user mailing list : -------------------------------- I suggest you put bridge=yes into each interface. Then, when Asterisk bridges two channels, it looks for the possibility to do a native bridge (call the bridge code of the channel

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

hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vvvvvcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _________________________________________________________________ MSN Hotmail :

The Asterisk Development Team would like to announce the release of Asterisk 13.30.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.30.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.7.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

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The Asterisk Development Team would like to announce the release of Asterisk 17.1.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

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The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Asterisk 18.0.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 18.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi all, I've a problem: I update my asterisk to version 1.4.25, and the attended transfer doesn't work. A call B, B press *2 and voice announce to digit internal and select internal of C. ---- CORRECT ---- A hear music on hold and B talks with C. ---- CORRECT ---- If B press *0, the call return to A. ---- CORRECT ---- if B hangup, ...... also the call hangup Someone can help