similar to: timeout music on hold or ring tone

hi all! Is it possible to trigger some external device trough parallel port or. serial on some caller id ? or if it's possible tu run some external (linux) program on specific caller id ? OR if anyone have idea how to open the doors with asterisk :) by, Thomas

hi! anyone has a idea how to send caller id (incoming number) to shell script like: exten => s,1,wait,2 exten => s,2,System(/bin/script callerid-number) exten =>s,3,answer i need to run script in shell with command line like "/bin/script -a 123456789 " this 123456789 must be callerid (incoming call) number. tnx, Thomas

hi, i installed asterisk and works very well, the only problem is that when i try to call a direct number of a company that has a normal PBX i got this error: to 10.8.210.153:5060 == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro) -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack -- Goto (doisdn,00115601992,1) --

hi! when i have busy signal on analog line (zap card) it doesn't detect that line is busy ? is it possible to change detected sequence (frequency) of busy tones on line (zapata.conf ??) tnx, Thomas my zapata.conf [channels] language=en context=lin1 signalling=fxs_ks channel => 1 group=1 echocancel=yes echocancelwhenbridged=yes rxgain=3.0 txgain=3.0 busydetect=yes

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

is it possible to do with asterisk something like this how ? maybe some copy of extensions and some other file ..pls. call to asterisk server -> user 100 is busy -> recorded msg say "all lines are busy , pls wait" ,"you are second caller in 'queue', pls wait" -> caller is on hold till user 100 busy then ring user 100 (ext100) another call to asterisk

hello, i have a problem starting asterisk with one x100p digium and two hfc chipset isdn cards with bri-stuff.0.0.2. "ztcfg -vv" shows me a this info: Zaptel Configuration ====================== SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03:

Hello, I got this information from the manufacturer of my ups Kobian/Mercury and were tested wtih powstatd on Linux (Works OK). Now i want to configure it with NUT and i didn't have any luck (it always says OB) Initiliasation for monitoring: RTS = 1 (High) Main power fail: CTS = 0 (Low) Batter low: DCD = 0 (Low) Kill UPS: DTR = 0 (Low) How would the ups.conf look like? Reagards, Lep

hello, anyone has worknig ISDN hfc-pci card in DDI (DID) point2point mode? what kernel ? and second question mISDN driver .. anyone has working solution with mISDN and maybe fritz card? what you suggest for DDI -> point2point mode (card,kernel,chan_..., ...) ? thank you, Tomaz

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that

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

Hey I need another pair of eyes on this! I would like to add phones numbers to the blacklist from any handset so I did this: exten => _*66XXXXXXXXXX,1,StripMSD,3 exten => _XXXXXXXXXX,2,DBPut,blacklist/BYEXTENSION/1 exten => _XXXXXXXXXX,3,Hangup However what I get in the database is: /blacklist/BYEXTENSION : 1 And BYEXTENSION is not replaced with the actual number

Hello, I just got my isdn-card working together with i4l and asterisk. Everything seems to be working fine: I can accept calls coming from the outside and I can dial out. Even setting the msn works like charm but my problem is that I cannot hear a word. There's complete silence in both directions. Any idea what could be the cause? Thanks for your help, Gunther Lspci: 0000:01:07.0 Network

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

Hello-- I'd like to do a little processing on external phone numbers from within the asterisk pbx. Fairly simple stuff, but... devilishly hard to make it work so far! 1. I'd like to dial 9 to get an outside line. 2. If the number dialed after the 9 is 754XXXX, I'd like it to go thru unmodified. It's the only local number available here. 3. I'd like all 1 XXX XXX XXXX numbers

Hi to everyone! I am new to this list and i want to greet you all! I am trying to install Asterisk on a Fedora Core 3 Distribution with a Digi Datafire Micro V ISDN Card. I know that I have to use the ISDN4Linux so in modem.conf I set the I4l driver ,the msn and device. Modem.conf [interfaces] context=remote ;driver=aopen ; modem by AOpen driver=i4l ; isdn4linux - an

I can confirm this in a slightly different setting, but still using two-way sync between two dovecots. On e is 2.3.19.1 running on macOS Monterey, the other is 2.3.20 running in an alpine container on Ubuntu. Gerben Wierda (LinkedIn <https://www.linkedin.com/in/gerbenwierda>) R&A IT Strategy <https://ea.rna.nl/> (main site) Book: Chess and the Art of Enterprise?Architecture

It might have a noticeable effect on clients. I encountered (probably triggered by this in some way?) that I was unable to het the 'read' bit set in macOS Mail.app. Maybe (as I am doing HA with round robin) the Mail.app client got to one dovecot repository on one tcp connection and then on the other. Is there a reason why syncing tis move from new to cur is a bad idea? Gerben Wierda

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

Hi all, Here another guy working with ztdummy and having problems with music. MP3Player does not work for me both from a telephone entering through a passive ISDN-adapter as well as a SIP-client in the LAN. Ztdummy works with conferences. Here are my messages. I have seen several discussion threads in the mail-archive regarding this subject, but could not find an answer. Is this something I