similar to: Using SIP phone to dial out using ISDN ?

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

I have looked at matt's asterisk gui client at sourceforge. I am not a programmer by trade. The documentation there seems to be a bit lacking. Has anyone have the experience in installing the gui client and may perhaps have a how-to document available for sharing. -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text

Hi, all, Can anyone tell where can I find the perl library for manipulating 'ini files'? Thanks, kaiduan ______________________________________________________________________ Post your free ad now! http://personals.yahoo.ca

I have an asterisk installation connected to 2 isdn lines via an AVM C2 card. modules seems to load well, lsmod gives : c4 19588 4 b1 24192 1 c4 capidrv 28468 2 isdn 134604 9 capidrv slhc 7552 1 isdn capi 18112 4 capifs 6024 2 capi kernelcapi 46112 4

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)

Hello. If I receive a Phone call by ISDN or from SIP Provider, the Asterisk make some errors and the SIP Client don't react. The messages from Asterisk in verbose mode: er will net. Asterisk messages in Terminalmode: parse_srv: SRV mapped to host sip-ha.web.de, port 5060 Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown"

Hi, whenever I include a "Ringing" Line in some Voicemail Extension I get an error when a call from the outside (via ISDN) comes in, but it works when an internal (SIP-phone) calls the extension. Here is my configuration for testing: ------------extensions.conf------------ [isdnext] ; strep external "101", our number and leave only extension exten =>

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Hi, i want forward a call witch is comming over isdn (avm b1 witch i have) out to isdn (same card 2. b channel). The call is comming (one b channel open one is free) the forwarding is processed (snom 200) all seems correctly. Then the message that the b channels all busy, but so is it not. Forwarding to a sip phone works. Can anyone help me with that ? nicolas SNIPS: ------ ==

Hi, i want use both B channels on my isdn card (B1 ISA) but chan_capi open one channel and asterisk say 2. channel is busy. Must i use another isdn card ? I have a old B1 ISA card. Can anyone help me with that ? nicolas SNIPS: ------ ??==?DISCONNECT_IND?PLCI=0x201?REASON=0x34a2 ????--?CAPI[contr1/<outgoingmsn>]/78?is?busy ????--?CAPI?Hangingup ????--?removed?pipe?for?PLCI?=?0x201

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

Is there a version of the pv_ops kernel that will do HVM and work with fc11? I have been failing miserably trying to get it to work. I have followed Boris''s guide at: http://bderzhavets.wordpress.com/2009/06/10/setup-fedora-11-pv-domu-at-xen-3-4-1-dom0-kernel-2-6-30-rc6-tip-on-top-of-fedora-11/ But have had no success so far. I am wanting to know what is the version of the pv_ops

asterisk-users-request@lists.digium.com is believed to have said: > >Hey Dan!! > >Give us a clue as to what hardware/setup & network provider you have there, >and we might be able to help :) > >Paul > Hello Paul, hello everybody! I have, too, an inbound call problem. I am using an ISDN Fritz Card PCI 2.00, together with chan_capi 3.5.x . As I call my number I get

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

Hello Armin, > The card is telling: > > CAPI INFO 0x34a2: No circuit / channel available > > so the other channel must be in use by something else. > Maybe another device on the ISDN line? > I have tested it several times now and always entered "capi info" before and after the call. The answer was always: Contr1: 2 B channels total, 2 B channels free. I'm

Hello, I'm glad to announce that DeStar 0.2.2 version has been released. This release contains a large number of bugfixes and new features, see CHANGELOG.txt for the full list. You can find it in the usual place: http://developer.berlios.de/project/showfiles.php?group_id=2112 Thanks for using DeStar, Santiago Ruano Rinc?n http://destar.berlios.de -------------- next part -------------- A

Hi all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time -------------- next part -------------- An HTML attachment was scrubbed... URL: