similar to: VOIP Router

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

If anyoane have a firmware with sip support for a tainet venus 2804 please give am feedback caz i kan-t find on internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100226/42f09275/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:

Dear All : I need to mask the caller ID and pretend to make a transfer call from another extension : exten => 558,1,Answer exten => 558,2,Playback(soundclip) exten => 558,3,Dial(SIP/472@callman) The scenario is like this : Someone is calling 558 at my company - he will hear a soundclip voice message then I will direct it to extension 472 I need 472 to not see the extension of

I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten => 558,1,Answer exten => 558,2,Playback(message.wav) exten => 558,3,Dial(SIP/439@CallManager) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error :

Hi, How often do you all have your ATAs and phone register with the asterisk server. I am doing it once an hour, but now I am wondering if maybe that is too long in between registrations?

I'm using a cisco 1760 with a VIC2-4FXO card for my calls to PSTN. If a user on a softphone hangs up first the PSTN port on the cisco is released and new calls can be made on the same voice port. But when the user on the PSTN side hangs up first the voice port on the cisco stays open until the user on the softphone hangs up. Any ideas what I'm doing wrong?

Hello, Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent Bleep (a private P2P SIP-based messaging application in early alpha) http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized-communications/ I have personally been a fan of Asterisk and have been using it for years and now that we have (kind of) released Bleep, I wanted to ask you guys to let

Hi Junaid, Sorry about the confusion, indeed I gave you the wrong output. So let's start to the beginning. I disabled quota and I reactivated it My configuration : Volume Name: venus Type: Distributed-Replicate Status: Started Number of Bricks: 2 x 2 = 4 Transport-type: tcp Bricks: Brick1: ylal3020:/soft/venus Brick2: ylal3030:/soft/venus Brick3: yval1000:/soft/venus Brick4:

Hi folks - I am configuring a new system and the antispam plugin is the last piece I need, everything else is working. Thanks to Johannes for this plugin, it's exactly what I want and an elegant solution for filter training. But I've been trying everything I can think of for the last 3 days to get this to work, no success. I've got: Postfix 2.5.3, dspam 3.8.0, Dovecot 1.1.6,

Hi, I have a R script file with Persian letters in it defined as a variable: #' @export letters_fa <- c('???','?','?','?','?','?','?','?','?','?','?','?') I have specified the encoding field in my DESCRIPTION file of my package. ... Encoding: UTF-8 ... I also included

Hi, I have a question about using RWeka Clusterers.If you could supply answer or insight, I would really appreciate it. When I run a simple code which uses a clusterer from RWeka I get an error. the sample codes and errors are mentioned below Code: library(RWeka) Cobweb(iris[,-5],control=NULL) Error: Error in names(class_ids) <- nms : 'names' attribute [150] must be the same

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

Centos 5 Hello I successfully converted my mta from sendmail to postfix. no problem. mail is ok. A by-product of that, as bizarre as it may seem is this: As regular user, when I do 'su -' to become root, all is well. As regular user, when I do 'sudo bash' I become root alright but I also get: sendmail: fatal: Recipient addresses must be specified on the command line or

My NFS server is on 192.168.10.10, with the following setup in /etc/exports: /backup 192.168.10.0/24(rw) Then, form a client (192.168.10.11), I run mount 192.168.10.10:/backup /bck, but get the following error: root at vps01 [~]# mount 192.168.10.10:/backup /bck mount.nfs: Input/output error On the main server, I see this in /var/log/messages: Mar 8 08:42:52 venus kernel: kjournald starting.

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

Dear all, As the next version of XiphQT is mostly ready, I thought it could use some more wide pre-release testing. The major change since last release is the addition of Ogg exporter and Vorbis and Theora encoders. Any feedback on how this new functionality performs (or doesn't!) with audio/video editing/producing software will really help. Also, comments and suggestions on the work of

Dear all, As the next version of XiphQT is mostly ready, I thought it could use some more wide pre-release testing. The major change since last release is the addition of Ogg exporter and Vorbis and Theora encoders. Any feedback on how this new functionality performs (or doesn't!) with audio/video editing/producing software will really help. Also, comments and suggestions on the work of