similar to: ISDN2 facility code...

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

Is there a way to specify multiple email addresses in voicemail.conf for a specific user? I seem to remember that it was possible, but can't remember the character to separate the email addresses. (I tried '&', but that didn't work...) later, PaulH

thankyou both very much for your swift responses and helpful insight... While my knowledge of administrating Asterisk is fairly decent, i must say my knowledgebase and ability in troubleshooting it is fairly lousy... all these wonderful suggestions you have had about turning this log on here, etc sounds great, but i don't know where to begin on that! how do you recommend i turn these on or

I need to carry a variable over into the 'h' priority - so I can go back and clean up DB entries in a mysql database (time of call and so on) I tried using UNIQUEID but it seems that 'h' generates a new one. Anyone have any ideas? What can I use to carry a variable over into 'h'?? later, PaulH

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales

Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice.... :) Doug.

Hi, I explain what I want to do.. All the operators share their phones. The number of the operator isn't constant, so it's possible that two operators share all the phones. They need to move all around, so they pick up the first phone they find. If there are only few operator is very annoying for them to ear the other phones ringing while they are at the phone! So I'dd like to limit

Does anybody know of any IP phones (ideally SIP based) that have interfaces to plug into a pro audio system (eg for phone interviews). Something can probably be hacked up with a headset connector or the 1/8" jacks on a 7970 but I'm wondering if there's something better out there. Thanks, Julien

I'm guessing that I need more info entered into the 'message centre' section. What did you key in? Paul Hales IT Support Adairs -----Original Message----- From: Jeff Pyle [mailto:jpyle490@gmail.com] Sent: Friday, 17 September 2004 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 I have two IP 500's on my Asterisk

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Which is the best interface card to connect PSTN line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show

I'm just getting started, and getting quite a strange result. I've been able to mount my (login hal) Linux home directory as E: on my w95 system, but when I try to "smbmount //isdn2/AMDAHL-WORK /mnt/amdahlwork", where /mnt/amdahlwork is an empty directory on Linux, isdn2 is defined in /etc/hosts as 192,168.1.2, the address the W95 box uses, and AMDAHL-WORK is the name under

I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code

Hello all What I'm looking for is some plain speaking advice on ISDN. Currently using 4 analog lines connecting via a four port TDM400P FXO card. We need to physically move our installations, and on requesting the analog lines be moved - our telco (BT) is suggesting we replace our analog lines with ISDN2. We would have 3 x ISDN2 connections, giving us six voice channels. They've

I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget

Hello! I am using Asterisk as transparent voice recorder for calls (isdn <-> asterisk <-> pbx). Voice recording (therefore voice forwarding) is working great but seems that Asterisk does not route/bridge/forward D-Channel messages which means PBX cannot get time synchronization answer from provider and tarification impulse too. With direct connection PBX works great and use both