similar to: broken pipe in perl agi

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

Dear, I am getting this in CLI on release candidate version of Asterisk. Any ideas, or points where to look? -- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi [Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe -- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0 Best regards, Josip

Hello List, I have been working on a little PHP software that uses AMI's UpdateConfig command in order to modify some of it's config files. I was working with 'Asterisk 1.4.22.1' and everything was working. After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type: ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe

Hey Guys, We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? extension.conf exten => 7770,1,agi(allpage.agi) exten => 7770,2,meetme(7770,dq) exten => 7770,3,playback(beep) exten => 7770,4,hangup following is agi debug....

All; I wrote a PERL AGI script that prompts a caller to leave a message using print "RECORD FILE $recordfile wav # 60000 BEEP s=3\n"; When the caller is done, they need to press the # key. The message is then delivered. However, the message is not delivered if the caller simply hangs up when finished. If the user hangs up, the script ends right then. How do I keep on processing the

I have upgraded Asterisk from 1.4.22 to 1.4.30 and I have noticed I am getting a lot of errors like this on the console :- ERROR[23912]: utils.c:968 ast_carefulwrite: write() returned error: Broken pipe I have tracked it down to a perl AGI script which performs our own CDR recording. It is called before the start of the call, once answered and again when the call is hungup. It works fine when

I recently upgraded to asterisk 11 from 1.8. I had VXML working via AGI in 1.8 - from extensions.conf: [VXML] exten => s,1,Answer exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})}) exten => s,n,AGI(agi://localhost/url=${ENCODED}) exten => s,n,Hangup Using asterisk 11 on the same host with the same config in extensions.conf: -- Executing [s at VXML:1]

Hi List I am having trouble running the command siptest:~# asterisk -rx 'dialplan reload' most times it does what I expect and I get a response as below siptest:~# asterisk -rx 'dialplan reload' Dialplan reloaded. every now and then I get no response i.e. siptest:~# asterisk -rx 'dialplan reload' siptest:~# and a "verbose 10" setting shows [Mar

Hi All, I'm having problems getting results from a PHP file. This is what the CLI is showing. -- Executing [111 at internal:1] AGI("Console/dsp", "GoTalk.php") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php [Dec 14 11:57:25] ERROR[20260]: utils.c:1019 ast_carefulwrite: write() returned error: Broken pipe If I run the PHP file from

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

## Hello.. I have asked a similar question, but this is not fixed as before. ## I am running the following using Ubuntu OS: R version 2.13.1 (2011-07-08) Copyright (C) 2011 The R Foundation for Statistical Computing ISBN 3-900051-07-0 Platform: x86_64-pc-linux-gnu (64-bit) ## when I do this: > install.packages("mapproj", dependencies=T) ## I get this: Installing package(s)

Hi All, This is my first post. I searched the archives and found something similar and I tried some of those suggestions. I changed the file permissions on the scripts directory to 777 (which doesn't seem secure), I also manually ran the detectdahdi.sh script. The response is "None". I am running Mac OS X 10.5.7 with Asterisk 1.4.26.1 which I compiled from source. The Asterisk Gui

Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly

Hi, After upgrading a debian/lenny server to 1.4.26 I get this error: == Manager 'munin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'munin' logged on from 127.0.0.1 [Jul 26 17:45:12] ERROR[12354]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe repeated each time munin logs in. Should I be concerned?

2008/11/4 Igor Zamocky <asterix at ponozky.sk> > > Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers > ? > I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is much more concise. Thanks a lot. > > > Hi, > > > In this

Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft

Hi, In this thread http://lists.digium.com/pipermail/asterisk-users/2008-October/219592.html , I wondered whether SIPPEER curcalls was working. I could test this anew today. Here are my findings : Alice, Bob and Carol ar all using SIP Phones. Whenever Alice is calling Bob, - if Carol is calling Alice, SIPPEER(Alice:curcalls) equals 0 - if Carol is calling Bob, SIPPEER(Bob:curcalls) equals 1

Here is what I have in my extensions.conf file now. Trustrcid and sendrcid are turned to "yes" in the conf file. I'm not fully sure how the SIPCalledRPID works though. The example I found seems to try and provide the stuff automatically (id and name), but does the SIPPEER stuff even exist? I think this is probably the right track though. Any insight would be much appreciated.

Greetings, I am writing an AGI script that needs to check on the idle/busy status of a number of SIP peers (mostly SPA9xx phones, with a few Polycoms and Snoms thrown in for fun). Is it possible to call Asterisk functions (e.g. SIPPEER) from AGI scripts? Based on my Googling, I would guess in the negative. I have tried various permutations of Set() and Eval() without success. I have also