similar to: Patch to enable syslog logging

Hi all, Has anyone already written something which allows saving and loading the internal DB settings? All users CFWD and speeldial settings are stored in the DB in my setup which makes it a pain to restart Asterisk.... Looking at showtree in db.c (why isn't that exposed in the CLI?) It shouldn't be too difficult, but I don't want to reinvent the wheel. On the same track, I am also

Hi, Has anybody seen a non commercial, or freeware, or GPL, or even CHEAP... POP/IMAP to Text-to-speech? I have a working version for POP3 using festival. It DOES work... it even cleans the email contents to get the actual content. It works great with Outlook emails and similar, and skips non multipart/alternative (that would be mainly SPAM, where the email is just

Simple Q but I can't find the answer in the archives (and am too lazy to look in the source, but then its 32 Celcius here... Do all digium cards provide the zapata timing? e.g. also the analogs (including the X100P) or only the E1/T1 -ones or do I need to use ztdummy on the analog cards? Thanks, Michiel Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen

Some switches provide the functionality to try a number till it becomes available. Thus when one dials a number and get a busy, one enters a *XX# code and the switch will call your extension when the called party becomes available. Has somebody already built this in/for Asterisk, otherwise I'll look into it. Michiel Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT

Hi all, I'm using asterisk CVS-08/14/03-22 on a box with a digium T1 connected to a channel bank and a digium E1 connected to the PSTN. I get occasional warnings from asterisk: WARNING[37909]: File chan_zap.c, Line 3197 (zt_read): zt_rec: Unknown error 500 This happens mosttimes in a loop like this: [netland_helpdesk] exten =>

Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel

I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

About every hour I see the yellow alarms on all or a number of channels of my PRI which is connected to the dutch telephony network, Asterisk keeps on working fine.... Here's an example where channel 1-24 went into alarm: WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 1: Yellow Alarm WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event):

Hi, We are using threewaycalling & flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but.... What if the

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting

This looks like a fun box... a Voip to Dect gateway, I've mailed them for pricing details.... < <http://www.computex.com.tw/news_archive_detail.asp?index=4053> http://www.computex.com.tw/news_archive_detail.asp?index=4053> Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen E: <mailto:info@betel.nl> info@betel.nl The Netherlands W:

I just downloaded the latetst CVS. A compile now complains about a missing srv.c & srv.h used in chan_sip.c. Can they be added? -- Betel Consultancy Abelenlaan 19 1185 RT Amstelveen The Netherlands http://www.betel.nl tel. +31 621 858 469

Hi, I would like to be able to log call details in Asterisk. The kind of logs that I like to generate is like this: From To Forward Time Incoming Call 604-343-3334 503-233-4454 13:33:32 Extension Routing 503-233-4454 Extension 403

I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel --

I have been unable to install Microsoft Office 2000 under wine on several Fedora Linux machines. The problem seemed to start with the June 2004 release of wine and has persisted for every version I have tried up to and including the latest 12/01/2004 version. I have tried both RPM install and compiling from source with similar results. I have one machine running the May 2004 wine that

Is case anyone wants to know... The Fritz! USB ISDN box works fine with Asterisk! I'm running CAPI 0.3.0 and love it, because the mini ITX server I have only takes one PCI slot which is now filled with a 4 port Digium card.

I'm trying to install Office XP on Fedora 3, with the 12/1/2004 wine build from the source. I tried the suggestions given by Ronald Nave on Dec 2/2004 (installing DCOM98, windows installer), but to no avail, and I get the following errors, any help/suggestions are appreciated! ------------------------------------ fixme:wininet:FindFirstUrlCacheEntryA ((null), 0x7fd943b0, 0x5b41ea38): stub

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

I was reading here and there about the differences between Merb and Rails [1] and became curious about statements such as this: > It is threadsafe with configurable Mutex Locks (Routing is also > threadsafe) Having mucked with the routes system, I had this awful feeling that I don''t understand threading well enough to know if my contribution is "thread safe".