similar to: how to adjust volume

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

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

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

Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

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

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

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

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

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

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

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.

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'm trying to figure out which would the best implementation for my mail and web server. Any experiences? Caveats? I'm on CentOS 3. -- <<JAV>>

Correct me if I'm wrong here but when a message bounces and the mailer/mta generates a bounce message shouldn't the from field have <> in it instead of an email addres (ie. MAILER-DAEMON@domain.com). The list was nailed with over 13,000 bounce messages(and they keep coming) from ONE list subscriber and I wasn't sure if their smtp server wasn't doing what it should. I

Hopefully this isnt TOO general of a question. I have a raid server that runs dovecot and exports via nfs the maildir format store. I have 4 mail servers running exim that use dovecot to deliver over nfs I have been examing our entire setup and had the thought, would it be recommended to instead have another exim process running directly on the raid server and have my mail servers deliver mail

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

On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith [SMTP:akohlsmith-asterisk@benshaw.com] wrote: > On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: > > Received: from source ([81.56.129.44]) by exprod5mx8.postini.com > > ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT > > > > Your MTA claimed it was called "SOURCE" but rDNS tells the recipient

I recently implemented v2.0.9 using ''shorewall setup guide'' 2004-07-31. Starting with block everything not known to be in use and opening ports as complaints come in. This has led to a few rule changes. After a rule change I use shorewall restart to reload the rules. Seems to work OK... except for an outbound NAT SMTP connection from a mail server on .122 to postini.com. The

1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/74328b51/attachment.htm

are there any open source sip softphone (Window OS version )?