search for: app_amd

I'm guessing this may be a question for dev list, but wanted to try my luck here first. I'm trying to compile app_amd (Answering Machine Detection) against 1.2.7.1 and am getting some errors. I should point out that I simply snarfed app_amd.c from http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714 ...so if there are other includes and such that are required, that would likely be the problem, but I...

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

...o test the applications as many as possible before submitting the code. However, I can't test all of them, so I would like to make a priority and list. If you want to use your favorite application with this feature, please reply this. I will add to the list. :) Btw, I'm considering the app_amd, app_queue for myself. :) For more detail of this feature, you can see it here. http://lists.digium.com/pipermail/asterisk-dev/2019-April/077270.html Thank you. Kind regards, Sungtae _______________________________________________ asterisk-app-dev mailing list asterisk-app-dev at lists.digium....

...the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26 NOTICE[28319]: app_amd.c:134 isAnsweringMachine: AMD using the default parameters. -- AMD: initialSilence [3500] greeting [1500] afterGreetingSilence [300] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [5] silenceThreshold [256] -- AMD: HANGUP any help is highly a...

Hi - I have been developing an auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c and while it works well for some phones it is proving to be very difficult tweaking the settings to get it to work reliable enough to go to production. If anyone is using this successfully in a production environment I would really appreciate any posts of settings you are using. My settings are a...

...seconds has already ended. It doesn't move to the next priority (which is checking the AMD_STATUS). Executing 'show channels' shows that the calls are stuck in the AMD app. I have another identical server where it doesn't happen. Both server have version 1.2.24 installed, and the app_amd.c source is the same (checked it by using diff). Recompiled the server many times already but the problem still exists. Does anyone else have the same problem? What version of app_amd is recommended for asterisk ver. 1.2.24? -------------- next part -------------- An HTML attachment was scrubbed....

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote: > > I get the desired use case to run app_amd from within a Stasis > > application, but I’m not sure about app_queue. You have everything at > > your disposal within ARI itself to replicate all of the functionality > > of app_queue and beyond. > > Yes, there are certain applications which are logically building blocks t...

...message and when particular message is found call should be hang up and dialplan should continue. So far we have come up with two ways we think solve the problem and would very much appreciate to hear your opinion. 1) add special progress extension to dialplan so we can start something like AMD (app_amd.c) as progress arrives to decode the message and than eventually call HangUp. 2) change app_dial.c to return to dialplan when progress arrives and than either call HangUp or something like WaitForConnectedCall. Right now we are using asterisk 1.4, but there will be need to switch to 1.6 (because...

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

...: Manager received command 'Command' Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command' Jan 24 17:01:45 DEBUG[31447] manager.c: Manager received command 'Command' Jan 24 17:01:56 VERBOSE[31455] logger.c: -- AMD: HANGUP Jan 24 17:01:56 DEBUG[31455] app_amd.c: Got hangup The amd.conf: [amd] initial_silence = 3500 greeting = 1500 after_greeting_silence = 300 total_analysis_time = 5000 min_word_length = 120 between_words_silence = 50 maximum_number_of_words = 5 silence_threshold = 256...

...so..). Hmm.. And here I'm stuck. Has anybody some idea for me where I can start looking for further solutions? Any help would be appreciated!! :) Thanks in advance!!! Best regards and greetings from Austria! johnny P.S.: Some detail infos: I'm using Asterisk 1.2.9.1. I installed app_amd from http://www.freedomphones.net/files/app_amd2.c My dialplan context for generating the outgoing calls looks like this (I use AEL): context 1000_amd_tests { _0043[1-9]. => { // be verbose.. :) Verbose(0,InfoInfo ${CONTEXT} - ${EXTEN} called on ${CHANNEL}.); // set our callerId to the...

...ping of backslashes occurs with MySQL (Reported by Javier Acosta) * ASTERISK-20850 - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality. (Reported by Diederik de Groot) * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported by Frank DiGennaro) * ASTERISK-21038 - Bad command completion of "core set debug channel" (Reported by Richard Kenner) * ASTERISK-18708 - func_curl hangs channel under load (Reported by Dave Cabot) * ASTERISK-16779 - Cannot disallow unknown form...

...200 ok received (Reported by Salah Ahmed) * ASTERISK-28625 - Playback of local files impacted by large media cache (Reported by Kevin Reeves) * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell) * ASTERISK-28608 - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) * ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL) * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when...

...200 ok received (Reported by Salah Ahmed) * ASTERISK-28625 - Playback of local files impacted by large media cache (Reported by Kevin Reeves) * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell) * ASTERISK-28608 - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile) * ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL) * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when...

Guys, is there any way to detect voice when calling a zap channel? For example, if you want to send out or playback a recorded message, you need to wait for somebody to actually answer the phone before playing starts.. Anyway to detect this?

...kind of trunks do you have for your outbound calls? (BRI/PRI/analog POTS/SIP/IAX etc.) I'm using PRI and it works very well - the dialplan doesn't execute the message playback until after the call has been answered. I don't know of any analog lines that can do that. (However, there is app_amd.c that tries to detect an answering machine vs. human answer which might suffice for dialing on analog lines.) BTW, the wiki has some nice dialplan examples that I pasted right into my extensions.conf file and I was working in no time. Please let us know if this helps. -MC -----Original Messa...

...k attempts to generate hangup event (Reported by Abhay Gupta) * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry Svyatogorov) * ASTERISK-27981 - res_fax: Fax session leak with fax gatewaying (Reported by pasandev) * ASTERISK-28419 - app_amd: Does not work with silence suppression (Reported by Nasir Iqbal) * ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure (Reported by Guido Falsi) * ASTERISK-27994 - PJSIP: Early media ringback not indicated after Progre...

...variable (Reported by Michael Maier) * ASTERISK-26006 - Show offending IP for TLS setup failures in logs (Reported by Oleksandr Natalenko) * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors not logged (Reported by Bernhard Schmidt) * ASTERISK-28419 - app_amd: Does not work with silence suppression (Reported by Nasir Iqbal) * ASTERISK-28018 - IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate (Reported by vijay kumar) * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when...

...y George Joseph) * ASTERISK-28392 - The no-partial-inlining flag isn't passed to the bundled pjproject or jansson builds (Reported by George Joseph) * ASTERISK-28402 - res_pjsip_registrar: SEGV in registrar_find_contact (Reported by Ross Beer) * ASTERISK-28143 - app_amd: Infinite loop on silent calls (Reported by Abhay Gupta) * ASTERISK-28353 - stasis: Crash at shutdown when statistics enabled (Reported by Joshua C. Colp) * ASTERISK-28374 - latest asterisk unconditionally launch gcc --version, even if the compiler is different...