similar to: Wierdness in AGI file

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<

Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten => 5150,1,Answer() same => n,Set(CHANNEL(language)=en_AU) same => n,AGI(testagi.sh) same => n,Hangup console output: -- Executing [5150 at AllPhones:1] Answer("SIP/PBX-00000024", "") in new stack

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

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

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

I have a SIP phone calling an AGI application. It starts out this way: -- Executing [s at macro-Call-AGI:2] AGI("SIP/151-b414f0c8", "computer-temp.sh,darwin,") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/computer-temp.sh Then I get a dozen or so copies of: [Nov 30 22:40:03] NOTICE[28300]: channel.c:2962 __ast_read: Dropping incompatible voice frame

When I use an eagi script when play a message appear a lot of warning messages, but it play very well I?m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300

Hi Guys, I have already tried this one on the developers list. I have not been successful getting much back there and I have notified them that I will post this on the users list instead. Hopefully somebody have tried something similar and can help out. I am developing AGI scripts on Asterisk and have run into some very strange behaviour and I think this is a bug, but I am not completely sure.

Every so often, I have an asterisk 1.4.22-4 system that goes into an endless loop with the following: [Jun 1 13:30:44] VERBOSE[13160] logger.c: -- Playing 'dir-nomatch' (escape_digits=) (sample_offset 0) [Jun 1 13:30:44] WARNING[13160] file.c: Failed to write frame [Jun 1 13:30:44] WARNING[13160] file.c: Failed to write frame [Jun 1 13:30:44] VERBOSE[13160] logger.c: -- Playing

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

I am using outgoing call files. I typically see the "ooh something changed / timeout" on a regular bases every second to be exact. Then it stops until some other call event happens. So I "mv" my call file to the outgoing spool directory, I am listening to that message, another call file is "mv"'ed into the directory and something happens to the timeout that its

So, per the discussion last week and generous donations, we have some new sound files with which to work. The sounds are located in: http://www.loligo.com/asterisk/sounds/ For those of you who just want to download the _new_ sounds, please fetch: http://www.loligo.com/asterisk/sounds/20040117.newsounds.tar All of the sounds in that tarball are also in the main ../sounds/ directory in

Hi, Happy Thanksgiving to those of us in the USA... Been trying to debug an AGI (in C) on 1.4.26.2. I blind transfer a call to this snippet of dialplan: exten => 999900,1,DeadAGI(pq.agi,999950) pq.agi then plays a prompt (which I hear just fine): [Nov 26 02:42:47] VERBOSE[28721] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/pq.agi [Nov 26 02:42:47] VERBOSE[28721]

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which I have missed? Or could someone give me hints on how I could implement this in the res_agi.c The

-- jand. more than just a group Asterisk AGI Command GET DATA is usually of this form GET DATA timeout max_digits When I execute this command, I get only a single digit, regardless of what the value of max_digits is, Also the script quits Immediately after the press of the digit regardless of what the value of timeout is, This is really un-desirable as I will like to GET multiple DTMF digits

Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307 struct timeval start; 2308 long sample_offset = 0; 2309 int res = 0; 2310

Hi, I've been noticing a lot of these messages lately: "NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?" Is something broken? I'm running asterisk-1.4.22.1. They seem to happen in a number of different places where a beep or recording is played, such as when someone leaves voicemail or when an AGI script I have plays a time announcement -- lots

Hello, im having trouble with the following: [Asterisk]<------>[ISP]<------>[ADSL Modem]<------>[Linksys Router]<------>[Grandstream ATA]<------>[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on