similar to: Reputable company for SIP/IAX2 trunking

Hi all, I've searched the depths of the resources on the internet; however I'm having trouble deploying a pxeboot solution via pxelinux. So far what I have successfully implemented in my infrastructure is a successful pxeboot setup for FreeBSD ( without the use of pxelinux ). I'd ultimately like to have a solution that will allow me to choose a network install of various Unix-like

Hi all, I'm looking for some rudimentary insight on GotoIf() which seems to be failing on me in my dial plan. All I basically wish to do is block a particular caller. Sounds easy enough, but my ternary operator/plan currently is not properly being implemented. Can anyone spot where I'm being a momo? All extensions get forwarded to the following macro: [macro-forward] ; arg1 = phone

Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any

Afternoon all, Is there any particular reason why yum fetches rpms for two architectures on almost any update/install I'd like to perform? This includes both i386 and x86_64. Here's a small excerpt after performing a 'yum update' cups-libs i386 1:1.1.22-0.rc1.9.18 update 107 k cups-libs x86_64 1:1.1.22-0.rc1.9.18 update 112 k

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Everyone I would like to use kicad for circuit design. It does not appear to be in the repos. I have built all the other software that I need and not included in the repos from source but I have struggled with kicad. There looks to be lots of places on the net that I can download RPMs not in the repos from but I don't know which ones run honest operations. Could anyone point me to a

Hi all, I have the following in my extensions.conf: exten => s,4,GotoIf($["${CALLERID(number)}" = "8585979857" | "8585970327"]?15:5) The numbers listed above are known spammer numbers. However, when I call from any other CALLERID, it still directs me to s,15 which is the Hangup() application. Here are logs from the asterisk CLI: -- Executing

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

Hi all, I've compiled zaptel drivers and reconfigure asterisk afterwards from source --with-zaptel. Modules are loaded accordingly: asterisk-1.4.2 # lsmod |grep z Module Size Used by ztdummy 5472 0 zaptel 194504 5 ztdummy crc_ccitt 3521 1 zaptel my musiconhold.conf: asterisk-1.4.2 # grep -v '^;'

Hi I am wanting to install : dovecot 1.0-stable I currently have dovecot 0.99.11-1 which came packaged on Fedora Core 3 It is doing auth off a MySQL db an working well I have downloaded the latest stable tarball and extracted ... I don't see how to enable mysql lookups at compile time... Help please ... I am getting a lot of pressure from the people at work ... Regards Andrew Andrew

I know that Rails 3 ideal of unobtrusive javascript will result in the removal of all inline javascript. But where does that leave RJS? It looks like PrototypeGenerator is still a part of the Rails core, so I''m going to assume that RJS templates will remain unchanged. Still looks like gems/plugins such JRals will be necessary for use of other JS frameworks. Can anyone confirm or deny

Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing

Hi Is anyone successfully using iax2 trunking with CVS head ? The reason I am asking is that I have heard there may be some audio problems, which I would like to know about before sending customer's calls over a iax2 trunked connection. Thanks in advance. Clive

I've just stumbled across a rather weird problem and was wondering if someone could shed some light on the situation. In testing faxing through Asterisk using Voicepulse Connect for trunking I am able to receive faxes without a hitch. Quite impressive considering previous experience with certain other VOIP providers. Today I finally got around to testing outbound faxing and found that if

Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Geo ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

Hello! I have bin trying to set up trunking between some of my Asterisk boxes but had no luck... I use Asterisk 2.0.6 on SuSE 6.2, but I have had the same problem with erlier releases. I have a working connection and can place multiple calls in both direktions. Than I set "trunk=yes" on both sides and reload. "CLI> iax2 show peers" shows a (T) and low latency on both

IAX2 now has support for a "trunk" mode ("trunk=yes" in the appropriate friend section). Trunk mode allows IAX2 to use bandwidth extremely effectively. The original impetice (and strategy) was a result of a mistake in which it was claimed that Asterisk with a T100P could send 96 simultaneous calls over a single T1 using VoIP. Thanks to a suggestion by the customer, combined

Hello, Would like to use IAX /IAX2 to transport 30 channels inter Asterisk. I don't have any experience with that, so can someone help ?? How much bw do I need for simultaneous calls and is there any latency for SIP G711 to IAX2 and vice-versa , ... etc ? Thanks in advance for any info, Geo ___________________________________________________________________________

I originally wanted to answer with something ... tzarit and kevit Readed probably before you invent rapid biz. I am asking to share any info /experience not your high spirit. Thanks for less trivial answer, G >On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote: >> I am using the inter asterisk trunking and the article in >> voip-info.org<http://voip-info.org>will

Greetings, I was hoping to get some feedback on a plugin I wrote. The plugin is called conflict_warnings and is currently available from my github repository at http://github.com/EmFi/conflict_warnings The purpose of the plugin is to provide a simple methods of preventing data inconsistencies that could arise from multiple users interacting with the same resource. Under basic operation a