similar to: Having Trouble Creating an IVR

Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around

Running fc3 with current cvs-head... Is there a nice way to rotate the /var/log/asterisk/messages file without shutting down asterisk? I'm currently rotating the log files via cron, however my script requires asterisk to be shut down, which also kills any outstanding cli sessions (eg, asterisk -rvvvvv). Would like to rotate the files without killing the cli session. Any reasonable way to

Im trying to use pxelinux-4.02 + memdisk as follows LABEL voyage-current MENU LABEL ^C. voyage-current LINUX memdisk INITRD voyage-current.iso APPEND iso raw console=ttyS0,115200n81 panic=5 pause LABEL voyage-0.6.5 MENU LABEL ^6. voyage-0.6.5 LINUX memdisk INITRD voyage-0.6.5.iso APPEND iso raw console=ttyS0,115200n81 panic=5 pause It mostly works - memdisk

can i use MP3 files as an IVR prompts directly without converting to .gsm format? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091010/67205e07/attachment.htm

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

Dear all, I wanna create a GNU/Linux that it boots from USB. I don't any idea on that. Can you introduce me a doc that i can use it & i can solve my problem? Cheers, -- ------------------------- Mohsen Pahlevanzadeh email address : mohsen at pahlevanzadeh.org web site : http://pahlevanzadeh.org IRC IM : m_pahlevanzadeh yahoo IM : linuxorbsd ----------------------------

I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the

Hi, I want to set up a Linux as Access Point, and maybe, as router too. There is some distro or minidistro in order to do that ? thanks in advance andres

Hi all, I have successfully compiled and installed Asterisk on an Alix board (AMD Geode 500 Mhz + 256 Mb RAM) on top of Voyage linux (a Debian variant). I'm using it at home for a month. I wondered how much it could be loaded, so I tested it with pbx-test: I could place up to 15 simultaneous SIP calls before it got no more responsive. All in all a good, stable and cheap solution for home

Hi, Some odd behaviour here. A friend and I were talking tonight, and it seems we have both seen the same problem. We are both using aastra phones (I am using 9113is). We have a connection to and from providers via SIP and IAX. When I place a call on the local hold of the phone, and then pick them back up I can hear them, but they can not hear me. However, if I park the call,

Greetings all! I've got some really silly questions. I'm a technical guy, and I understand how the astrisk server works and how VOIP works, etc... The problem I have is that at my small company we have a phone system with analog lines and everyone here is comfortable with the concept of using them. I've never seen IP phones in action so I don't know how they work from a users

Hi Once I installed astrisk , how do we connect with SIP interface ? Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose. Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130511/a6d32bfe/attachment.htm>

Dear Members, I am searching for a new PBX for the company. My choice is Astrisk. My Boss wants background music via all the telephones. This is done in a conventional PBX that he wants, but I can use the Asterisk PBX if it can do this also. As I said he needs background music on every telephone this is not to be mistaken with music on hold. The bit stream is an MP3 file of 8 Kbs. At the server

Ever since upgrading to 1.2.8 I've been getting occasional WARNINGS that say: WARNING[4356]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed. Is there something I need to fix or is this a benign message? Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.394 / Virus Database: 268.8.0/352 - Release Date: 5/30/2006

Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI

>> Straight away the booting process looks different than with 4.06. >> The periods that were showing progress in loading the initramfs now >> does not increase as the initramfs is loaded. So, only three periods >> after each initramfs. > > That happened in version 5.00. Good to know that this is expected behavior. However, some sort of indication that the initramfs

I have question thats not really about astrisk but I figure you guys are doing this sort of thing. We use Aastra 6757i phones. there is some support for XML. the question is how would i go about learning to customize these phones? _________________________________________________________________ Bing brings you maps, menus, and reviews organized in one place.

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Hi.., we have two asterisk server interconnected to each other through IAX2 trunk in two separate office. with this bellow configuration do we need to have Licensing for using G729 codec???? Office A --------T1 ----- Astrisk TE05P----------------IAX2----------------Astrisk Box -2 | |

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891