similar to: SIP remote crash bug

AGI's are resulting in unusual behaviors. Can someone please tell me if this is my inappropriate use of AGI's, inappropriate use of Time::HiRes, or a bug with *: I call this script twice: #!/usr/bin/perl use Time::HiRes qw( gettimeofday ); ($seconds, $microseconds) = gettimeofday; $hirestime = sprintf("%s","$seconds$microseconds"); print "SET VARIABLE

Hi! I have configured a SIP trunk with a dialing rule. The trunk behaves normally for incoming calls but when in used for outgoing call a strange thing happens. When I place a call I cannot hear the tone confirming that the remote phone is ringing. I simply hear the voice as soon as the party picks up. When the remote phone start ringing Asterisk receives a SIP packet stating that the call is

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>

Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168>

Hello, I've got an Asterisk system with 3 SIP trunks configured. Each SIP trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound call routing (via AMP 1.10.007a) uses the 3 trunks in descending order, all set with max channels to 4. Unfortunately, when the first trunk reports a "480 Service Unavailable" (all ports in use), Asterisk reports congestion without

Guys, I'm having a hellava time getting realtime to work, focused on sipusers right now, followed the wiki and other examples but still no luck. Using mysql on a seperate server, asterisk actually sees the database and can poll the table "realtime load sipusers" at the cli but asterisk realtime engine is no pulling the user info. I'm using 1.2.4 stable and have the database

Hi, We are using VOIP-SIP gateway to route outbound PSTN calls. Recently, I am getting == No one is available to answer at this time message, after making 5 SIP attempts (Retransmitting #5 (no NAT):), and the calls are going out through alternate Zap-trunk. I do not see any hit (sip-debug traffic) on the voip-gateway for the failed calls. Strange thing is that this is happening randomly,

I setup my aah box with a sip trunk at irisxa.iristel.net Incaming it is ok but when I try to dial 8 and the nr where I want to call I get all line is busy. In my log I have these: Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:04 DEBUG[3156] manager.c: Manager received command 'Command' Feb 22 14:33:19 VERBOSE[2721] logger.c: --

Asterisk Project Security Advisory - AST-2009-005 +------------------------------------------------------------------------+ | Product | Asterisk | |---------------------+--------------------------------------------------| | Summary | Remote Crash Vulnerability in SIP channel driver |

Asterisk Project Security Advisory - AST-2009-005 +------------------------------------------------------------------------+ | Product | Asterisk | |---------------------+--------------------------------------------------| | Summary | Remote Crash Vulnerability in SIP channel driver |

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

Dear All, Just hope your guys out there can help me through..since I've been playing for serval hours....and still not able to resolve it... The workflow as I've created an .call file for Asterisk to call out and it's working fine with outdial, passing variable to asterisk..But the problem is when the calls reached Context and execute AGI script, the script didn't get any

Hello All. I am trying to use RAGI the ruby agi framework with trixbox. I am having a problem with the dialout part. The RAGI framework creates a file in the /var/spool/asterisk/outgoing directory and routes the call to an extension (I have listed the relevent portion of the file below). The problem is that the initial dial command does not execute properly in trixbox. I am hoping somebody who

Please see my extensions below. I will try to type you through this. In default, an extension 5149053538 is matched on. This fires a macro that determines what time is it, and resets a variable which is then used to call another macro to place a call. if the call is answered, and the far end hangs up, the dial macro exits, then the routing macro exits, and you are back to default. at this point

I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls

Barton Hodges wrote: > I've been messing around with a "free line" notification > where an extension is dialed for a second when a line becomes > available. I can't seem to get the "h" extension to continue > when the local party hangs up. I've seen references to other > people having the same problem in the list archives, and the > solution

Hi Guys, I'm having sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone

Hi Folks, I'm have Asterisk version 1.2.1 with a A101 PRI card. I'm working with the CLEC to bring up the PRI and inbound calls are hanging up at his end after a few seconds. I ran PRI debug but it only gives me minimal insight. " Ext: 1 Cause: Unknown (16), class = Normal Event (1)" He ran a trace and the only difference he is seeing is a "ISDN interface explicitly

I am using Festival to synthesize some menu Interaction with a caller and am having a problem. What I am working on is a remote callback where I can remotely call in to an extension, and enter a callback number (or use the CALLERID info) and a second outbound dialing number to connect to. Things work O.K. until I set usecache=yes in festival.conf. After doing this, things run well for

Is there a limitation to the number of variables you can set from a PHP agi script? I have a simple example and I can't get it to let me set more than 1. I am pretty sure I am just missing something, but I've searched all over an can't find the answer. Here is the extensions.conf part: exten => _XXXXXXXXXX,1,AGI,diallocal.agi exten => _XXXXXXXXXX,n,NoOp(${ISLOCALCONTEXT})