similar to: Pre-pending certain digits (like 9) to an outbound call number

Hi all, So I'm trying to work on this complex fax server setup, and part of it involves connecting my asterisk server to my Rolm CBX switch, via a T1 line. I plan on using Asterisk simply as a T1-PRI Bridge to IAXmodem (which in turn, activates HylaFax+ to handle the faxing). So far, though, I don't think I'm getting 100% of the way there. When dialing the fax extension from my

Okay, T1 card issue sorted out. New Lesson: Stay Away from TigerJet chips. Next up, modem pool -- I wanted to know if the below config looked anywhere near half-sane for defining in asterisk what is essentially a small pool of four waiting modems that will handle faxes if another modem is busy: exten => _X.,1,Dial(IAX2/iaxmodem0/${EXTEN}) exten => _X.,2,Busy exten => _X.,3,Hangup

Hmm, I don't know if the zaptel fax detection will trigger. This is going through my company's Rolm CBX switch, using a plain T1 cable (yanno, RBS, E&M Wink, and all that jazz) between the two systems as a Tie line, so that may mess with the fax stuff. The Rolm's just wired to take a special extension block and pump anything coming in on them out the T1 trunk group to the

I am currently a new asterisk user I have worked with the old rolm systems in the past. I have been asked to look around and find out how to do a few things in asterisk, either in asterisk itself or with third party software. The features that I am looking for are: 1. A good management application for setup and edit the .conf files for both the exten & voicemail. 2. A receptionist software

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

i've heard that you can dump a plot file, and in doing so set a pixel parameter (cbx?). is anyone familar with this procedure? i would appreciate any insight. thanks, kevin

What an ACTIVE newsgroup! I'm in the early stages of researching Asterisk. My current environment is a small college (~1000 sets/~400 student sets), Avaya Definity G3si/Seimens Rolm Phonemail. As you can imagine, the maintenance, licensing, and equipment costs are HEFTY. So.. are there any small colleges/universities using PBX or Voicemail? If so, I'd be interested in your migration

Hi all, Noticed a curious issue in my testing setup for a faxing system I'm putting together, but it looks like if I let the lines all sit idle for a few days (no one uses this yet, so the whole thing really does sit idle until I do testing on it or something else), something I believe on my Asterisk end goes to a kind of "Sleep". It's hard to describe really, but I'm not

I have found that the shortcut I add to the windows machines has a few options added to it for it to run. eg: T:\Acucorp\Acucbl620\AcuGt\Bin\wruncbl.exe -c lawpac.cfg lawstart.cbx T:\ is the network mapped folder that is accessible after logging on to the servers with Novell Client, this folder may also be seen on suse using the client. the shortcut location is here: H:\TREVOR\LawpacWindows

Hi all, Having fixed my problems with the call waiting ringing on the GS phones, I needed to extend that with a campon facility (available on some legacy systems - sort of callwaiting without phone ringing). I've managed to implement that by adding/modifying app_queue.c. Basically, when calling the SIP phone, and if busy, I can camp the call onto that extension, with MOH, allowing the caller

I am currently working on an Asterisk test system, and will be presenting a demo to the Board of Directors tomorrow night. I want to make sure I have all of my ducks in a row. The Asterisk system will be used to replace a Norstar MICS. The location has two PRI's coming in, with a few hundred DIDs. I know how to make * use the DIDs incoming, and I know how Nortel uses the DIDs. Now for the

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

Im trying to figure out the base way to check the callerID being sent to my Asterisk box and use it if it is a valid NANP number, but replace it with a static NANP number if it is not. (Why? I have a few carriers that require this, and a few international users - if it happens to take one of the carriers that require it, I want it to set a static number that is valid). I'm playing

Hi, I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while to fix all the deprecated stuff, but everything seems to be working fine now, except for a little tiny thing. I lost all color in my CLI, which makes it harder to debug. Is there something that needs doing? I didn't explicitely disable colorization from the command line, and I did try using nocolor=no in the config files.

Does Asterisk work with Lucent or any other PBX phone systems????

Are there affordable DIDs (preferably IAX) available anywhere outside the US? I want to use it to meet ICANN requirements for providing a valid phone number, yet pre-empting some of the telemarketing calls my domain registrations generate. (Yes, I asked a similar question about 900# availability before). I'd prefer to have a number somewhere outside the NANP, preferably an asian country.

Apologies for the off-topic question; as usual I'm trying to draw upon the unparalleled knowledge and sagacity of the r-help list. Please reply off-list if you can help me out. A collaborator of mine found a formula we need, on sheets which he had photocopied out of a book, some years ago. He cannot remember which book (he's getting to be as senile and forgetful as I am, poor bloke!).

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

Hola, What would a valid regexp in Asterisk be to identify a NANP number, i.e., NXXNXXXXXX? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:brent.torrenga at torrenga.com web:www.torrenga.com

I run a SIP-only shop with a 23 channel PRI and single T100P. Here are my configs: /etc/zaptel.conf: span=1,0,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us /etc/asterisk/zapata.conf [channels] context=default switchtype=dms100 signalling=pri_cpe pridialplan=unknown rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=no