similar to: 705 DIDs for Collingwood Ontario?

I believe that there is one more patch required for a successful build on Tru64: # diff ./src/lib/mountpoint.c.orig ./src/lib/mountpoint.c 60a61,63 > #ifdef __osf__ > point_r->type = p_strdup(pool, getvfsbynumber(buf.f_type)); > #else 62c65,66 < point_r->block_size = buf.f_bsize; --- > #endif > point_r->block_size = buf.f_bsize; Cheers

Hi folks Setting up a new server, hence a few software packages have been upgraded / changed. On my old server (only 2 years old) I was using Dovecot and RoundCube (web mail app) and had a good experience so was keen to use these again. The only major change is I have switched from sendmail to exim4 (which was installed by default on this Debian server). Exim4 appears to be doing the right

I have recently written a (minimal) Tru64 Unix SIA password module for Dovecot as part of testing a Dovecot installation. Has anyone else written a Tru64 Unix SIA module? Is anyone else interested in such a module? If so, how might I/we go about getting this/such a module into the main Dovecot source? Thanks -- Simon L Jackson Carringbush.Net +- Carringbush.Net Hosting * Development *

Hi, Is there a way of limiting the number of IMAP connections (i.e. processes) per user? Despite asking users to set their email client settings to limit themselves to 2 connections, some still have this set to 10 or more. Cheers -- Simon L Jackson Carringbush.Net +- Carringbush.Net Hosting * Development * Advice Web: www.carringbush.net <http://www.carringbush.net/> Email:

Hi, We have a mail server with multiple instances of Postfix each listening on a different IP address to create a set of virtual mail servers. There are various reasons why separate instances of Postfix seems to work best to achieve this goal, but I wont divert into that discussion here. I would like so match this with Dovecot. The listening side isn't a problem as Dovecot can listen on

Hi everyone, I googled this followed the instructions, but it hasn't work for me yet. I have universal setting in SIPDefault.cnf and phone specific settings in SIPXXXXXXXXXX.cnf. But it doesn't get registered. I need to register it on two different asterisk boxes. So my SIPXXXXXXXXXX.cnf looks like this: phone_label: "Zeeshan A Zakaria" line1_name: "523"

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)

We are testing the innomedia ATA's to possibly replace our current line up of ATA's that we are using. Has anyone used their product? What is their track record on stability, voice quality, DTMF talkoff, T.38 Thanks Bryant ---------------------------------------- From: "Zeeshan Zakaria" <zishanov at gmail.com> Sent: Wednesday, October 13, 2010 10:41 AM To:

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

For a while now I've been looking for a good web based soft phone solution, but so far no luck. A few solutions which I've tried, both Java based and Flash based, either don't work, or had bad sound quality. I need something which I could put on my productions server for my clients. Seems like good web based solutions are all paid ones, nobody is giving it for free. Any ideas,

We had very similar problems to this which drove us insane for ages. Basically we use VoIP trunks (SIP) for all our inbound + outbound calls. Call quality was good however we would get random problems where people could not hear us or us hear them for about 5-10 seconds at a time. After weeks of trying to get to the bottom of the problem it appeared our VoIP trunk provider (sentiro/sip2go) had

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice

Hi list, For a client I am setting up a system which will use T1 PRI from Primus, who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have only used switchtypes euroISDN and National. Although the documentation says Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you have used these protocols on an Asterisk box and if there were any things to consider. If

For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to

Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

Hi everybody, How can we add new contexts in asterisk realtime module? All I could figure out after googling is that a new context HAS to be declared in extensions.conf with 'switch => Realtime/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for

I've got a 778 DID for vancouver, but don't know if it will be a local call if dialed 604 and vice versa. What are the different area codes in Vancouver and why its easier to get 778 DID than 604? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

Murray Collingwood put forth on 4/20/2011 8:29 PM: > Hi Stan > > I only have the one account on this server, everything else is relayed > elsewhere. If this is an SMTP relay server, I'm surprised you bothered to install an IMAP server at all. It would have been easier to simply alias postmaster@ to another account on another machine where you do store mailboxes. I.e. simply