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hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

Hi there, I'm tring a basic asterisk settings. I have a asterisk 1.2.7.1 running on a I have a net with two computers and a router. The router IP in the local net is 192.168.1.1, The first pc has IP: 192.168.1.3 name datile3 . SO GNU Linux. the second pc has IP: 192.168.1.4 name fissun . SO GNU Linux. On datile3, it runs a softphone kphone. From this I want to call the external world. on

AstriCon Update: Only Two Weeks To Go! October 12 - 14, 2005 Anaheim, CA AstriCon 2005 starts two weeks from today. We now have a complete roster of speakers covering Asterisk from soho to carrier. We've added the Code Zone, a working lab with a full compliment of VoIP and TDM equipment. We also have over 20 confirmed exhibitors and more are joining the event each day. AstriCon 2005 is

Hello, I get some problem installing asterisk + ekiga on my debian etch: ii asterisk 1.2.13~dfsg-2 Open Source Private Branch Exchange (PBX) ii ekiga 2.0.3-4 H.323 and SIP compatible VOIP client $: asterisk -U asterisk -vgc give me some WARNING like : ,---- | WARNING[21806]: res_musiconhold.c:852 moh_register: Unable to open | pseudo channel for timing... Sound

The agenda for Astricon Europe in Madrid June 15-17 is now coming together. It will be an international conference, with speakers from both USA and Europe. Last year, we had over 25 nationalities participating in the first Astricon - the Astricon where Mark released Asterisk 1.0 on the conference floor, during his keynote! Many active members of the Asterisk community talks at the conference, one

Thanks to help on this list and Sangoma's support we have incoming and outgoing calls passing through asterisk. However both incoming and outgoing calls are greeted by silence. I've noted our existing config below with our test extensions.conf. Help much appreciated Rory Zaptel ----------------------------------------------------------------------- loadzone=uk defaultzone=uk #Sangoma

Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/1001 at from-sip|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten

hello, I have instaled a analog line, and when I call on the console apears that: I want to redirect the call to 101 extension. *CLI> -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at default,s,1 still failed so falling back to context 'default' Apr 27 08:15:53 WARNING[3494]:

The AstLinux Team is happy to announce the release of AstLinux 0.7.5 with options for both Asterisk 1.8.1.1 and Asterisk 1.4.36. More information about the release is available on our website: http://www.astlinux.org/content/astlinux-075-release Direct links to the installation files are available here: http://www.astlinux.org/release/075-asterisk-1811

Hi All, Today on the VUC, we'll be welcoming Sipgate CEO Thilo Salmon back to tell about their choice of partners in their latest services. I will be announcing the first #VUC VoIP & Tell discount code for Astricon in Denver, October 25-27. Join us on sip:200909 at login.zipdx.com using g722 or g711 codec or see http://vuc.me for the other ways to connect. Hear you there! /r

Hi I'd like to allow users to transfer a call to a specific park number. This way, the receptionist can tranfer a call park for ext 100 at park number 7100 etc... It seems like this should be fairly simple using the Park(ext) app but it doesn't work for me. No matter what I extension I use, the system just picks the next available park number. I've simplified my dialplan for

The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash "ShellShock" bug. New in 1.2.0: * New Linux Kernel 3.2.x * "igb" ethernet driver for Intel Atom C2000 * Enable AES-NI support * New "sip-user-agent" firewall plugin * New versions of Asterisk 11 and 1.8 * Bash "ShellShock" security fixes A

The Astlinux Development Team is happy to announce the release of AstLinux 0.6.7. This release is a security and bugfix release with no new features. All current users of AstLinux are encouraged to upgrade. Current users can upgrade either from the web interface or by issuing the following commands from the CLI. upgrade-run-image check http://mirror.astlinux.org/firmrware (which should

I'm trying to connect an Asterisk system to an Avaya Partner ACS R6 system. The problem I'm having is that I cannot get the partner system to get CallerID over the T1 modlue. The partner is using the T1 with E & M signalling (which I don't think can be changed), and whatever I tried didn't work. My only option right now is to get FXS ports on the Avaya side plugged into the

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

Hello everyone, Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems with high loads (and probably any system at that) - especially if it is finely tuned and has zaptel

Hello everyone, (I first sent this several hours ago, it appears it got lost. Thought I'd give it a second try). Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems

The latest version of Astlinux has been released today. Changes include: Asterisk 1.4.24.1 (security fix AST-2009-03) Asterisk-gui svn 4739 Astlinux web interface 1.4.07 Update Zoneinfo to '2009e' Additional checking is now done when performing an upgrade of the OS. Existing users of Astlinux (0.6.x) can upgrade their system from the web interface or by issuing the following commands

Hello everyone, I have just finished up work on a new release of AstLinux, the embedded/live cd/minimal built from scratch Linux distro centered around Asterisk. Most of the work for 0.2.8 has been on the ISO image (live/install cd), the init system and further system customization with new rc.conf variables. - The ISO image now allows you to install AstLinux to a local disk/CF card,