similar to: FAST BUSY on Back to back ZAP outgoing calls

Are you using freePBX by any chance? Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nigel Roberts Sent: 08 November 2006 08:55 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue forks asterisk and then leaves the extraprocesses lying around Hi, I have a problem with Queue

Hello, I'm using Asterisk@Home. I'm still new to Asterisk, and trying to grasp it all. I'm wanting to do a simple setup of One SIP provider (Broadvoice) and 3 SIP Software Phones. I'm able to call the VoIP provided line fine and get dropped to the digital receptionist (or mailbox). However, when I try to send outbound calls I get "Error 503 Service Unavailable" and

Hello, To do some testing with Asterisk installed the latest Asterisk @ Home in a Vmware system. All worked fine, I can access the web interface (AMP). I have setup the extention and X-Lite softphone according to the description in the Wike (http://www.voip-info.org/wiki-Asterisk+phone+xten+xlite). I can dial 200 (the softphone extention) and 1234 and they connect (the softphone shows this, as

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Hello, I have an Asterisk @ Home box up and running. I'm trying to prepare the box to be the phone system in our new office. I have 3 Polycom 500's with the latest SIP firmware and all is fine, I can call Asterisk, check my voicemail, call the other handsets, transfer call, even make calls to the PSTN via voipjet. So I'm doing ok. The problem I have now is that when I reboot

I have an application where I need to send outbound prerecorded messages. The Asterisk "call file" process works fine if I am sending the call via SIP or IAX, but not via ZAP over a PRI channel. The destination device (my cell phone) never rings. The only unusual thing I see is on the fifth line of the "full" log, below, "channel.c: Don't know what to do with

Hello, I am having an issue whereby calls are being dropped randomly. I have an ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk install is based on Trixbox 2.0. However, I have updated the source code to the following. The Asterisk release is asterisk-1.2.20. Zaptel release is zaptel-1.2.18. And libpri release is libpri-1.2.4. I have include an extract from the Asterisk log

> > Hi, Remove usecallingpres=yes busydetect=yes from your zapata.conf file. and the restart asterisk. Hopefully you will not faced drop call issues. Regards, Vidura Senadeera. Message: 3 > Date: Mon, 24 Sep 2007 12:29:40 +0100 > From: "Richard Young" <Richard.Young at intrintech.com> > Subject: [asterisk-users] Asterisk Dropping Calls > To:

Hi, I'm currently running CVS-HEAD 2005-09-03 I do plan to upgrade to the newest version, but need to do some testing with it first. In the mean time... does anyone know what these messages below are about? I've never seen it before, but when it happened it locked Asterisk up pretty good. Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on Feb 10 10:16:57

Hi a new small question ;=) We have two Asterisk, connected in IAX2. On the first, in dialplan, we have: exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) we sent into the IAXVAR "ACCOUNTID" the accountcode. On the second, in dialplan, we have: exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) That's work, the second server get the variable. I

I'm having a problem with * (tried both HEAD and STABLE). When logging with cdr_odbc through unixODBC to MySQL, I get "insane" integer values in the duration, billsec, disposition and amaflags fields. I have enabled MySQL logging, and that's the query I get: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the database structure for cdr_mysql is: CREATE TABLE cdr ( calldate datetime NOT NULL default '0000-00-00 00:00:00', clid varchar(80) NOT NULL default '', src varchar(80) NOT NULL default '', dst varchar(80) NOT NULL default '', dcontext varchar(80) NOT NULL default '', channel

Hi. here is a brand new cdr_mysql . Is based on the previous one, but with lots of changes, so I include here the whole file, not a patch. What I've changed: added new fields, in order to reflect cdr csv : * call start time * call answer time * call end time * call unique id Changed the table structure to reflect cdr.h lengths. Added some (a lot?) sanity checks, to be sure to insert

Hi everybody, Firstly I'm going to describe the scenario where I'm working. I use a E400P with Asterisk CVS-05/22/03-11:14:50, and I'm working with asterisk trow AGI scripts (Perl). The configuration of extension.conf is: exten =>_s,1,Answer exten =>_s,2,AGI,script.agi Inside the AGI script is call Dial application as follows: print "EXEC Dial

I noticed a problem this morning with our cdr logging. We have a cron job that places a call file into the spool directory having asterisk call itself to check to make sure its still handling incoming calls correctly, then queries the CDR database in mysql and makes sure that appropriate records exist. I can confirm that the call is happening correctly, but I'm missing records in the