similar to: Dial application timeout

hi all i am using CVS with Realtime mysql on backend. Dial application is invoked again and again what is the reason. I have tested it by printing some message to debug. this application is invoked again and again here is debug you can see lot of messages from app_dial.c at the end. Any one tell me what is the reason. Is this a bug or what Kamran Ahmad

Hello, I am trying to do : Exten =>_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:10000)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:10000)' Why? I forgot something ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que

Hello friends, I am a newbie to asterisk , please help. I am receiving a phone from a sip server and I want to route it to another sip server. The problem is that the target sip server takes a # in the argument . I am trying to dial as exten => s,5,Dial(12345#123456789@xxx.xxx.xxx.xxx) ;;;; dial the number at the ip address. When this gets executed, I get the following eror:-

Hi All, I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial

I am trying to setup an extension in asterisk which dials an extension on the NEC DSX. i.e. If an asterisk user dials 402 I want it to connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. ( 404 would be the NEC DSX sip account that I have the credentials for ). [402] deny=0.0.0.0/0.0.0.0 secret=pass1 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic type=friend

Network Configurations Block D, Surrey Park, Barham Road, Westville, 3610 Helpdesk: (086) 163-8266 Tel: (031) 266-1563 Fax: (031) 266-4206 Hi people. I'm hoping someone has come across this problem with version 1.2.14 In my dial plan I call various SIP phones using the following little macro: exten => s,1,Set(TIMEOUT(absolute)=14400) exten =>

Hey, I'm having some errors whenever I dial out and I can't dial in at all. I'm using NuFone as my provider just so you know. Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: No such device Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unable to set device to input mode Jan 11 17:39:46 WARNING[1771]: app_dial.c:359

After updating to 1.4.15, I have the following issue: When I try to use the "M" macro option in the Dial() option, I get the following in the console: -- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT") -- Called g2/w5051234 -- Zap/3-1 answered Zap/1-1 [Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to start

http://lists.digium.com/pipermail/asterisk-users/2004-October/065540.html I saw this post about the M(x) option for the Dial command, but I could not find a reply questions posed here. I am wanting to pass the Zap channel that the original call came from to my macro embedded in the Dial command. I've tried to add arguments to the macro by using the syntax M(x,arg1), and I always get the

--- man/man1/unicorn.1 | 32 ++++++++++++++++---------------- man/man1/unicorn_rails.1 | 34 +++++++++++++++++----------------- 2 files changed, 33 insertions(+), 33 deletions(-) diff --git a/man/man1/unicorn.1 b/man/man1/unicorn.1 index 0b496af..749272a 100644 --- a/man/man1/unicorn.1 +++ b/man/man1/unicorn.1 @@ -4,7 +4,7 @@ unicorn - a rackup-like command to launch the Unicorn HTTP

Hello, I wonder if there is a way to build my own asterisk application (let us say apps/app_myappl.c), and to launch other existing applications from it (for example, doing an apps/app_dial.c, or others). Could someone highlight me on that? thx Pirlouwi. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, With a 1.4.35 or 1.6.1.19, I'm facing this behaviour : - extension 7002 is a SIP hard phone currently configured to forward incoming calls to extension 7003, when a call is unanswered within a 10s time frame - when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20) statement and no one answers, then : - after 10s, Asterisk receives "SIP 302 Moved temporarily"

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

Hi All After a Dial() I get: WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) if the subscriber is not registered. Is there a way from dialplan to know, *before* Dial(), if a destination Subscriber is a) not registered or b) busy ? I need to redirect a call to some other Subscriber if (s)he is not there

Since I only seem to get questions, and no feedback, from the Wiki page, I'll ask here. There seems to be no lack of opinions here... I have a working wakeup call system on my home * system. The architecture is something I'm not perfectly happy with, though. There are two AGI scripts, written in Perl, which handle (a) scheduling, confirming, and cancelling a wakeup call, and (b) the

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Hi, I have an Asterisk 1.2 (can`t upgrade to 1.4 because of some makefile error on my particular system, bug report opened). That being said, I doubt my particular issue is a bug, I think it's me not understanding something. Let`s take a simple dialplan command, i.e. make the phone ring for 15 seconds: Dial(SIP/some_sip_registration|15) It works well (most of the time). If I disconnect

What I am trying to do is the following: A call is sent to the * box via a SIP invite. The * box answers via an IVR menu system with " Enter the extension you want to dial" so I enter in my 5 digit extension and get the below message. Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No channel type registered for 'SIP)' Jan 18 10:10:03 NOTICE[-1380238416]:

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1",