similar to: Update the LCD with the callee's name after dialing

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Sorry for maybe not a very list related topic, but I have always been curious if there is information on how many Asterisk based PBXs are operating Worldwide? Thanks and hope the community will not reject my curiosity! :) Best Regards, Vallu Sevana Oy -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101020/8ab7ae3e/attachment.htm

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

Hi, I know some of you are very experienced as to the working of networks. I wondered whether there is some accepted way of determining bandwidth needs based on the network traffic over time. For example, looking at the figures for the network traffic through the server interface, we have hourly, daily and monthly figures. If everything were linear, taking the hourly figure and dividing it by

Hello my friends, I'm very worry about a problem i'm having...my asterisk got freez some times, every 5 or 6 days with NO trace in /var/log/asterisk/messages What i want to know is if safe_asterisk has something to be with this? This is what i have on my server: [root at mypbx ~]# ps -A | grep asterisk 9118 ? 00:01:30 asterisk [root at dreampbx ~]# ps aux | grep asterisk root

With previous Asterisk versions when running asterisk -r a welcome message is displayed with the version. I just upgraded to 1.8 and noticed it is not appearing. All I get is Verbosity is at least 3 and the console prompt. I looked at main/asterisk.c and still see the welcome message code. Any idea why it is not being shown? Ryan

Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside

I have two problems. The first is that when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. The second problem is that I thought I'd try an internal phone to see if I could get the hello-world stuff working at the least. I

Hi, On freepbx (GUI), whatever reason number fails we always get 'all circuits are busy' audio. Does anybody know how to get far end audio when we dial wrong number or when it's busy or unallocated number or failed with some other reason. Thanks, Deepika -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anybody know where to find some good information on the new CEL in asterisk 1.8? I'm very anxious to check out the new logging features but can't find anything but the cel.conf.sample file in the source package. I'd like to get this setup with ODBC. Thanks in advance.

Hi guys, a little OT but I figured this is the place that would know. Is there a free or paid webapp where I can get inbound sms messages? I only need to receive a few inbound sms messages a month but it cant be my current cell number :-( Any thoughts? Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, Can anyone please tell me how to use the phoneprov.conf to provision my client's atas. I read the file but dont know how to actually use it. -- Best Regards Rizwan Qureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101027/7f9de26c/attachment.htm

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

I have a setup with Yealink phones & Asterisk Server (all latest patches). I am using BLF to display the states of other phones. While this works MOST of the time (busy, being called) it does NOT work when a phone is NOT regisstered at all, the yealink phones display a green dot EVEN if a phone is turned off (try explain this to users, they are shaking their heads!!!) I can see on the

I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows "Forbidden", and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" <

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <small><font face="Helvetica, Arial, sans-serif">Hello,<br> <br> I have a question concerning SNOM M9 base station.<br> <br> If my customer places a SNOM M9 base

Hello list, This may or may not be Asterisk related, but if I had hair I'd pull it out over this. I have a TDM400P card in a Dell POWEREDGE 1550 running Asterisk 1.4.30. Everything works great except that every time it rains, I get flooded with this CLI message - == Starting post polarity CID detection on channel 1 -- Starting simple switch on 'DAHDI/1-1'

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I have come across the above handset a few times in the UK, They are quite cheap over here (~?80) Not the best handset in the world but works well enough. I have been asked to setup a central config server for a large collection of these handsets. I know they can do Auto provisioning via FTP/HTTP/TFTP I have got an example of the generic

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila -------------- next part -------------- An HTML attachment was scrubbed... URL: