search for: 2basterisk

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

2009/5/20 Gordon Henderson <gordon+asterisk at drogon.net<gordon%2Basterisk at drogon.net> > > On Tue, 19 May 2009, Olivier wrote: > > > Hi, > > > > In ASTDB, I've got a rather long list of entries like: > > /FamilyA/Key1 Value1 > > /FamilyA/Key2 Value2 > > /FamilyA/Key3 Value3 > > ... > > > >...

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz

Hi, I have a really strange problem with a Polycom 430 phone and Asterisk 1.4.20. Currently If I dial the Polycom from my mobile phone answer the call on the Polycom and then hangup the mobile the call ends fine on the Polycom. But if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the

Hy guys i am having so much hard time to setup asterisk on a virtual machine that i got , i just want to know if i really need to use Dahdi and libpri on a complete Digital PBX i just gonna use sip and iax. I will never use any kind of analog line on this machine. Wait for a feed back. Daniel Abreu.

Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Does anyone (other than cynical old me) think that Enum is a spammers best friend? Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) I can see that

Does anyone have a suggestion on how to handle this? For example, if I have a list of numbers that I want to go out a certain sip channel and another that I want to go out the dahdi device, is there a way to do this? None of the numbers will fit into a pattern, so just plain pattern matching won't do. The most straightforward way would be to just define explicit patterns. Obviously that

Hi Everyone, Knowing that running Asterisk on an embedded board like the Alix2d3 requires some fine tuning. Do you know of any good guides out there that does this from beginning to end? Looking to run this in a small office environment. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring

...mailto:tzafrir.cohen at xorcom.com > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir > > > > ------------------------------ > > Message: 18 > Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST) > From: Gordon Henderson <gordon+asterisk at drogon.net<gordon%2Basterisk at drogon.net> > > > Subject: Re: [asterisk-users] Can Asterisk bridge between a SIP client > and a Cisco Call Manager server? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <Pine.LNX....

Hi, How can I create a web page allowing people to listen (with their own PC) a couple of .wav/a-law files stored on a Linux server ? Chances are users would access this web page from Internet Explorer but if I could make it available to other browsers, that would be better. I googled a bit and couldn't find a tag such as media://myaudiofile.wav that would fulfill this spec. As much as

Hi all, I've done this a few times with other PBX's but swyx has stumped me! I'm having some trouble getting Asterisk connected to a Swyx system using a sangoma A104dx... currently the setup is: BT <-> Swyx The above setup works fine... what i'm trying to achieve is BT & SIP Trunks <-> Asterisk <-> Swyx I have connected to our BT (2 x ISDN30 UK) with

Hello ! I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it,