similar to: Comparisons between * and sipXpbx (PingTel's open source product)

I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. /carmi

I was just about to replace 2 Merlin Legend systems (one in my house and one in my Parent's house) with * systems. I have 2 beefy linux systems ready to go and have enough Cisco 7960 and 7970 phones to replace all my MLX-10D, MLX-16D and MLX-20 phones. Then I discovered that * seems to be lacking a critical feature for replacing a legend (or any standard PBX) and that is an ability for a

I am looking for a provider that will provide an equivalent of DID/DOD trunks via IAX, IAX2 or SIP using numbers in Metro Chicago (prefer Skokie), LA (prefer West Hollywood or Venice), and/or Orlando (prefer Winter Garden). If I can migrate some of my existing numbers using LNP, that would be even better, but it is not a requirement. While I know that there are several companies that will

1) Who bought Pingtel's phone line? 2) Anyone seen this chinese-made VoIP phone that supports 8 different protocols? http://www.telecom.globalsources.com/GeneralManager?language=en&design=c lean&action=GetArticle&article_id=9000000055338&page=printarticle&printT his=yes Mike :) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Slash-dot is pointing to this article on Asterisk and Pingtel. http://www.theregister.co.uk/2005/05/22/pingtel_voip/ Paul Paul Mahler www.signate.com

Hi all, I am running centos 5.1 and I wish to change the domain name and dnsdomainname of my PC. currently the settings are-- $ hostname sipx.com $ hostname --fqdn sipx.com $ domainname (none) $ dnsdomainname com I have searched in the net for tips but everywhere only the hostname change is provided. I need to change/set the domain name and the dnsdomain name on my pc to sipx.com and this

Hi I am trying to your wine with sipXecs. We are testing Cisco ATA-186. When we attempt to update the ATA software via sipX we send a text file to the sipX server but it needs to be convert from a text file into a binary via the cfgfmt.exe tool I installed sipX and wine on the same box and used wine config tool to point or grab the tool. Wine puts it or a copy of it in the system32 folder.

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

* Brethren, It's a sad day in our community. Please join me in a moment of silence for the death of responsible journalism. Silence.....................good enough. This article goes on to tell about Pingtel's announcement of forming the "first open source community aimed at creating SIP based servers".

Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running

Hi all, I want to deploy a SIP Proxy but I just don't know which one to choose. Researching in the Internet I found the following ones: * SIP Express Router <http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is used by many SIP providers standalone or in conjunction with Asterisk * Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org> * sipX

Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards, Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120103/ffad2be6/attachment.htm>

I understand that Asterisk 1.4 should support T.38 pass-through, but I need Asterisk (or something on the Asterisk box) to act as a T.38 endpoint. Judging from the unclaimed $12,000USD bounty, it doesn't appear that Asterisk itself can do this. http://www.voip-info.org/wiki-Asterisk+T.38+Bounty Does anyone have any experience with this, or are able to point to an example of this working?

Hello all, I have problems trying to register sipXphone to asterisk. It always print the message : May 26 13:04:33 NOTICE[2781]: chan_sip.c:7691 handle_request: Registration from 'sip:2503@172.25.50.52' failed for '172.25.49.219' Anyone has an idea to help? Asterik seems to fail in register_verify() but i don't know why... My sip.conf is the following : ; ; SIP

Hi all, I just watched the rail video: http://www.rubyonrails.org/media/video/rails_take2_with_sound.mov I am just curious. I can tell that the video was done on a Mac, but I cannot tell which applications were used for editing the code, and for interfacing with MySQL database. As far as editing source code, I normally use BBEdit (I''ve used it since 1994), but the editor used in

Hey there. I've been working on a project to integrate Asterisk with Exchange Unified Messaging via sipX using large parts borrowed from: http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html ... and everything works surprisingly well. The one problem I have is MWI, or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so I've been looking into

Hello All, I have pingtel and asterisk working really well. I have a really annoying little problem - mainly with pingtel. When a call comes in pingtel displays the caller ID on the phone. If I miss it then I click on the number for redial - this doesn't include a 9 to dial an outside line. The second problem is with the dialer from outlook again it bypasses the outlook dialing rules so

This Friday's edition of the weekly VoIP Users Conference call is all about wideband audio (aka HD Voice) and conferencing. The guest for this call is David Frankel, CEO of ZipDX a commercial service that specializes in wideband conferencing. We expect an interesting call touching on many aspects of VoIP going beyond the traditional phone service, conference bridges, technical standards,

I am trying to figure out is * can be configured so that users would have a similar experience to what they receive on a legacy PBX. For example, on a merlin legend (or similar PBX), I have phones that have shared call appearances so that my assistant can answer my calls or see that I am on the phone, or so that I can have one phone on my desk and one at my conference table. This means that

Hello, Does anyone have an up-to-date page describing, step by step, how to make a customized CentOS5 DVD? I noticed that CentOS5 already comes with ~240MB of updates. So for starters, I'd like to create a new DVD with all the current updates. (And I have other custom scripts I need to install on top of that). I've googled around and tried various suggestions on the net: