similar to: List delays

Hello list, I am getting the list with days of delay, take for example this message: Received: from unknown (HELO lists.digium.com) (216.207.245.17) by mxavas16.fe.aruba.it with SMTP; 29 Jun 2007 13:38:37 -0000 Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from <asterisk-users-bounces at lists.digium.com>) id

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

how to adjust voice volume for sipura 2000 and cisco ata186?

Any ideas? This one would make the 10th email received. Return-Path: <centos-bounces at centos.org> Received: from psmtp.com (exprod8mx192.postini.com [64.18.3.171]) by mail.pdinc.us (8.12.11.20060308/8.12.11) with SMTP id p2LKkur4025229 for <jpyeron at pdinc.us>; Mon, 21 Mar 2011 16:46:57 -0400 Received: from source ([72.26.200.202]) by exprod8mx192.postini.com ([64.18.7.10]) with

Hi guys, This should be has an easy answer for you, my users are complaining that when they press # and then ear gorgeous Allison "Transfer" the timeout is very small, they must enter immediatly the extension to transfer the call. Is it possible to change this? ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call This is timeout

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Every time I post to the list I get the following response indicating that there is a mail loop. the recipient maxx@digital-system.it has some problems with looping. Hi. This is the qmail-send program at smtp.register.it. I'm afraid I wasn't able to deliver your message to the following addresses. This is a permanent error; I've given up. Sorry it didn't work out.

My SIP seems to be configured correctly as I can dial out and my minutes show up on my broadvoice bill, but whenever anyone calls my broadvoice # inbound they just get a busy signal. I dont get anything in the logs saying anything came in from broadvoice at all. Has anyone had this/simmilar problem with inbound from Broadvoice? Any suggestions? Thanks Neri -------------- next part

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

Hi! I have a E1 PRI connected to my TE400P card on span 1, and two channelbanks on span 3 and 4 and * 1.2. Every few hours I get this message and asterisk dies just after that: Warning: No D-channels available! Using Primary on channel 16 anyway! When this happens restarting zaptel and asterisk services, generally puts the system back online my zaptel.con reads: span=1,1,0,ccs,hdb3

We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? /Why Tea

Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An

I can ping the google.com ip addresses (209.85.231.104) but can't ping the FQDN, obviously DNS is broken. I can also ping the nameserver ip addresses specified in /etc/resolv.conf. -- Gaurav On Sun, Mar 7, 2010 at 10:48 PM, Gaurav Nangla <gaurav.knangla at gmail.com>wrote: > Hi Everyone, > I've setup a Centos 5.4 (x86_64) box in a VM recently; the trouble is I just >

I have a mail server that handles several domains. One of these domains has decided to use Postini. For those not familiar with Postini, you set your MX records to use their mail servers. They filter mail, and deliver you only the clean virus/spam free mail. The idea is to only allow incoming mail from their mail servers so spammers are unable to send to your mail server directly. This is

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Hi, Is there a list of all the extension states as sent by the manager interface? (I know I could look them up in the source but that involves some "backtracing".) The ones I know are: -1: no hint for the extension 0: registered && idle 1: busy 4: unreachable, not registered 8: ringing I've recently seen 16 (== hold?) but can't find that value documented anywhere.

Awhile back I had heard some talk, in this list I believe that Agent callback login was going to be deprecated in 1.4, I see it is still there. Does anyone know what is happening with this? -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 voip at rockynet.com

Has anyone seen this before? These phones are behind an edgewater. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bKaf87f1c9f;received=xxx.xxx.xxx.xxx. From: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=23943befc9dc103 To: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=as2c0b7dcd Call-ID: 723559d029d27c820c8dae4b01e45c77@192.168.50.31 This phone is

Hi, Suppose I have a multidimensional array: tmp <- array(1:8, c(2,2,2)) is there a function out there that, given a one-dimensional array index, will return the separate indices for each array dimension? for instance, tmp[8] is equivalent to tmp[2,2,2]. I'd like to derive the vector (2,2,2) from the index 8. thanks, Brad Buchsbaum

how can I fix this just started ...... Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid