search for: waitforsil

I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. Some general info on my setup: more /proc/version: **** Linux version 2.6.16.60-0.34-smp...

I'be been having problems with the WaitForSilence application in my dialplan. The behavior I've noticed, and hope to demonstrated with the fallowing examples, is that WaitForSilence (WFS for short) does not work until Background is called. Note that the lenghts of the sounds are approximately 5 seconds for bring, and 10 for presses. [Te...

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is actual silence. If I'm still talking, it will consider that as silence also. My AEL dialplan associated with the calling is: > 100 => { > Answer(); >...

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is actual silence. If I'm still talking, it will consider that as silence also. My AEL dialplan associated with the calling is: > 100 => { > Answer(); >...

Hi all. I have a customer who does some automatic messaging.? Back when we were running Asterisk 1.4.x, they used waitforsilence after amd, to wait for an answering machine greeting to finish before leaving their message.? We've upgraded to 1.6, and made no other changes and things don't work. What we're seeing it that a call will get hung up, waiting for silence, well after any sane greeting message would h...

...= 5000 min_word_length = 120 between_words_silence = 50 maximum_number_of_words = 5 silence_threshold = 256 In extensions.conf [outboundmsg1] exten => s,1,NoCDR exten => s,n,AMD exten => s,n,GotoIf($[${AMDSTATUS}=AMD_PERSON]?humn:mach) exten => s,n(mach),WaitForSilence(2500) exten => s,n,Playback(outboundmsgs/msg1) exten => s,n,Hangup exten => s,n(humn),WaitForSilence(500) exten => s,n,Playback(outboundmsgs/msg1) exten => s,n,Hangup -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail...

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

...o mention that this is not at all for spamming, or telemarketing. This is an appointment reminder service. from extensions.conf: [mycontext] exten => 899,1,Answer exten => 899,2,Wait(2) exten => 899,3,AMD exten => 899,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach) exten => 899,n(mach),WaitForSilence(2500) exten => 899,n,Playback(were-sorry) exten => 899,n,Hangup exten => 899,n(humn),WaitForSilence(500) exten => 899,n,Playback(welcome) exten => 899,n,Hangup The call goes out fine. When I pick it up AMD basically locks up, although not exactly because as you can see below...

.../gafachi-081c81a8]. Too long... -- AMD: Channel [SIP/gafachi-081c81a8]. Too long... -- Executing [15155515509 at robocop2:4] GotoIf("SIP/gafachi-081c81a8", "0?human:machine") in new stack -- Goto (robocop2,15155515509,7) -- Executing [15155515509 at robocop2:7] WaitForSilence("SIP/gafachi-081c81a8", "4000|2") in new stack -- Waiting 2 time(s) for 4000 ms silence with 0 timeout -- Executing [15155515509 at robocop2:8] Playback("SIP/gafachi-081c81a8", "mr-roboto-short") in new stack -- <SIP/gafachi-081c81a8> P...

...extensions_table limit 0,3; +---------+-------+----------+----------------+ | context | exten | priority | app | +---------+-------+----------+----------------+ | 10 | start | 1 | Set | | 10 | start | 2 | AMD | | 10 | start | 3 | WaitforSilence | +---------+-------+----------+----------------+ Any ideas on where to begin w/ the debug would be very appreciated. Regards, Pedram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081105/0e0a8e...

...IAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten => dial,n,Agi(agi://localhost/Outdial.agi, ${DIAL_STRING}) exten => dial,n,Hangup() exten => answer,1,NoOp(Dial Status = ${DIALSTATUS}) exten => answer,n,Playback(${GREETING_NAME}) exten => answer,n,WaitForSilence(2000) exten => answer,n,Agi(agi://localhost/Outdial.agi) exten => answer,n,Hangup() Everything seems to work fine so long as the Dial command executes successfully. For example, if someone picks up the other end after the Dial command completes, processing jumps to the "answer...

...did... AMD didn't work at all on my release.. I > think I was using 1.4.11 at the time... > I ended up using the below > daveC > > ;-----------< amdtest (ext 13) starts here >-------------------- > ; > ; restructure this for the following conditions: > ; 13 using waitforsilence(variable set) then play message > ; only works when there is an answering machine picking up and it doesn't > ; cut off (hangup) before SILENCEDURATION ms > exten => 13,1,NoOp( Starting exten 13 AMD stuff) > exten => 13,n,Wait(1) > exten => 13,n,Set(SILENCEDURATION=4...

I am having a bit of a problem getting AMD to work on a new server. On my regular office server it works like a charm. I am running Asterisk 1.4.13, Zaptel 1.4.5.1 on both machines. Both servers run CentOS 5 and I am using a SIP trunk to send out calls (the same one on both servers). Here is the output of a call on my office server: -- Attempting call on Local/0445540881644 at CC2 for

I want to be able to take an existing pre-recorded audio file, and dump it into an extensions voicemail box. My application is that someone/thing has dialled an extension which has triggered an event - lets say it's turned a lightbulb on via a System call in the dialplan, and I then want to send the owner a voicemail message (pre-recorded) which says "Lightbulb is ON". Now, I

...dir to place the call call is connected to [extension] which contains a playback app.While line is ringing, playback is triggered I want to start playback file when call is answered to make sure that called party hears message from beginning. What's is different now. I try with Wait and WaitForSilence, nothing is fixed. Does anybody know how to fix that. Your help would be highly appreciated. -- -- Adolphe CHER-AIME Network Integrator CCNA, CCNA VOICE, Global VSAT Forum Certified (509) 3748-3875 / (509) 3449-4280 -------------- next part -------------- An HTML attachment was scrubbe...

Hi all, i'm trying to make AMD, Answer Machine Detection, to work on my outbound context but i can't get it to work, just on inbound context like whe i use the application Answer before AMD, but i need to make AMD to do the detection on an outbound predictive dialer integration. Follow are the inbound and outbound examples. My current environment is Asterisk 1.4beta3 and a Digum

All; I have an application that dials a list of numbers and then plays a recorded message. My customer uses it to dial a list of customers to confirm their appointment for the next day. No biggie, maybe 25 - 30 calls per day for customers who want the confirmation call. What they need now is a way to dial an extension after the number is dialed and answered. I've seen that before, but I

...application 'Math' [app_realtime.so] => (Realtime Data Lookup/Rewrite) == Registered application 'RealTimeUpdate' == Registered application 'RealTime' [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_waitforsilence.so] => (Wait For Silence) == Registered application 'WaitForSilence' [app_while.so] => (While Loops and Conditional Execution) == Registered application 'While' == Registered application 'ExecIf' == Registered application 'EndWhile' [app_setrdnis....

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,

...t; >> 1.6 is not on my radar. >> >> >> -- >> Thanks, >> Steve Totaro >> +18887771888 (Toll Free) >> +12409381212 (Cell) >> +12024369784 (Skype) > > Hello, > > We really just maintain a set of patches for 1.2 (just updated > waitforsilence a couple weeks ago in fact) and we regularly install > 1.2.30.2 in call center setups. It is rock solid and extremely proven > in high-call-volume situations. > > We have started installing 1.4.21.2 on some systems that are not high > load as well (1.4.22 has some strange issues...