similar to: Bug in voicemail module of Asterisk 1.4.2?

from extensions.conf: exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) I basically try to lookup the CLIP and attach a name for each inbound call. This works fine, except when I have just restarted asterisk - at which time I've more than once seen the message from the subject. As far as I can tell, with my Set(CALLERID), I should always have an argument in the DB

I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is

Anyone have a quick recomendation for a 2-port (ie. 4 channels) BRI (ISDN2e) card in the UK? My usual source has 1 port and 4 port, but no 2-port cards... Thanks! Gordon

Hi, New to Kubuntu and Linux, I'm looking for a syntax-enabled text editor with which I could easily edit Asterisk config files. It seems Kate provide this type of service but I couldn't find anything specific to Asterisk (unlike vim) What's your advice ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten = _X.,n,Voicemail(443,u) exten = _X.,n,Hangup() When I start asterisk, this queue doesn't work - -- Executing [4439000@Business:3] Queue("mISDN/3-u0", "enidan|t|||20") in new stack [May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable

I am new at this. I have read "Asterisk: The Future of Telephony" and have installed AsteriskNOW (beta 4, due to the dual processor problem in beta 5). The GUI interface does not seem to provide the capability that I need, although I have modified the *.conf files to successfully create what I need. Given this, I would like to install Asterisk on a distro. I am most familiar with

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

Over the last two weeks, we have had at least two "incidents" where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further).

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

I've asterisk stop (presumably segfaulting) a couple of times, and I was just beginning to look at how to keep it running - what have others done? I was thinking of wrapping a script around asterisk like this: while 1 do asterisk -f done /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

I've got a Grandstream 487 in a home-office. The phone-side is working fine, but the user is complaining that his internet connection keeps disappearing. The Grandstream is set up as NAT router, and there's just one PC hanging off the LAN. Has anyone experienced anything similar? /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

> On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson <oej at edvina.net > wrote: >> >> 7 feb 2010 kl. 15.09 skrev Per Jessen: >> >>> Thomas Winter wrote: >>> >>>> Hi, >>>> >>>> my Asterisk on debian lenny died after 80 days. >>>> >>>> server kernel: [7572666.186852] asterisk[3673]:

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

Since upgrading from asterisk 1.2.x to 1.4.18 I've noticed a change (bug) in the voicemail messaging emailing operation. I had set serveremail option to: voicemail at pbx.mydomain.com and under ast 1.2.x messages arrived at user mailboxes from voicemail at pbx.mydomain.com . However, since upgrading emails arrive from voicemail at firewall.mydomain.com (the path from asterisk box to

Hi there, I am using redhat 9.0 with asterisk 1.0.7. I created an user and was be able to leave voice messages to that user and retrieve the voice message. I looked the wiki and setup the voice message as the email attachment. However, I have never received email with the voice attachment. Here is the setting for voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats

Hello, I was wondering if there is a way to change the "From" address (not just the Return-Path) for voicemail notification emails in Asterisk. It looks like the serveremail directive in voicemail.conf just changes the Return-Path. I'm looking for something analogous to the -r option in mailx, for example. I need this since the mail server I'm using requires the sender to be

Hi, My Gigaset S450ip allows 2 simulatneous calls when each incoming call are targeted to different phones. When both calls target the same extension, the second one is forwarded to voicemail. I couldn't check yet SIP messages but has anyone met this limitation (one simultaneous call per phone) ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: