similar to: how to enable dial to alex@asterisk.blurb.com

Hello, all. With the assistance of very helpful folks, our brand new multi-tenant setup seems to be working smoothly from start to finish with just a bump or two. The biggest is parking. Now that we got most kinks worked out, I'm a little more comfortable in trying to resolve this. There seem to be two problems: 1. Parking assigns parking spaces from the default group no matter

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)

Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, "we can't," but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic

Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [root at pbx01 asterisk-1.6.1.1]# grep -i speex

Hello, everyone. No need to read this message. I'm posting for documentation for other poor, ignorant slobs like me who are struggling to pull together the many technologies to make converged networks happen. Hopefully, this will help save someone else the time I spent. I started the below email until I realized I had solved multiple parts of a compound problem but not all at the same time.

hi all i've just setup an * box and want to test voip calling, initially from sip user to sip user... local sip users can call each other, no issues. problem arises when i try and call a remote sip account, my * box always returns "SIP/2.0 404 Not Found" any ideas ?

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together

Hi Guys we have a system that uses a lot of custom hints based on the extension the extensions use the format of ext-system for example 200-pbx01 when starting asterisk the "core show hints" show the correct hints and blf works as expected in the extensions.conf we have _.,hint,Custom:${exten} when running dialplan reload all the hints lose the dashes (-) they become 200pbx01 of course

Hi, We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are unable to URI dial our clients. We run a multi-tenant server and have set sip.conf to forward calls to a public context based on incoming domain name. This was all working before but not it is complaining of a loop back as the source and target server are the same. Any ideas on how to overcome this problem as we dial

I recently began using the curl cmd to do an external callerid lookup on my own customer database. I've noticed certain lookups will cause a crash and not show anything in the messages file or the console. The curl command is connecting to an external webserver which has a oracle db connection. The file its hitting is PHP and does a very simply lookup showing the text like "C1234 Bobs

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

I''m using windows, if I find the time I''ll try to build it myself. Otherwise I''ll just wait for the next gem. Thanks for a great job on wxruby2 btw! Regards, Mattias > > > > Date: Tue, 21 Nov 2006 08:40:53 +0000 > From: Alex Fenton <alex@pressure.to> > Subject: Re: [Wxruby-users] TaskBarIcon i wxruby2-preview gem > To: General discussion of

I want to personally recommend Simon @ Simwood eSMS for any DID, SIP or IAX needs in the UK! Simon responded PROMPTLY and PROFESSIONALLY to my request to establish a DID for my interview on Jan. 26 with Alex Tew, creator of phenomenon MillionDollarHomepage.com. The only thing I needed to do was register with eSMS's server, and the interview commenced FLAWLESSLY -- the call was CRYSTAL clear

Quick reminder before Astricon (from which we will be reporting from live): Tomorrow's guest will be VoIP author Alex Robar. Alex has worked with open source telephony solutions for the past four years, and has collaborated on the development and growth of an international Asterisk-based VoIP peering network. His book is FreePBX 2.5 Powerful Telephony Solutions and we'll be chatting with

I will not have access to email on Wed 30th April 2014. If you need urgent support please email or call Dan Benton on dan at dogsbodytechnology.com or 07718 679512 Thank you Alex Last

Hello, I have 2 asterisk servers that are not working well together. One is acting like a registrar (PBX01) for all my PAP2's and other SIP/IAX devices. And the other is acting like my sip gateway (PBX02) to various providers. They are both on a private network and should be trusting each others IP 100%. But the PBX02 challenges PBX01's requests all the time even though

hi all, how to establish a call between two asterisk servers for the sip users registered for the servers. ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Sunday, February 10, 2008 11:30 PM Subject: asterisk-users Digest, Vol 43, Issue 30 > Send asterisk-users mailing list submissions to >

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