similar to: Any small colleges/universities using PBX or Voicemail?

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

Hey all, Working slowly on getting the myriad number of parts to my fax system plan together, and one of the pieces I want to nail is how to go about, for the outbound context (fax-out) pre-pending a digit onto a number? I.e., for all my testing right now, I've been dialing '91XXXXXXXXXX', as the asterisk server doing faxing junctions into my old Rolm CBX switch, and so I need 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

We have had quite a bit of success with our T100P and TE410P cards interfacing to Nortel Meridian PBXes and also to a Livingston Portmaster 3 using ESF/B8ZS and various combinations of E&M wink and ISDN PRI (usually in 5ESS mode). In the near future, I may also need to interface to a Definity via a T1. I was planning to use PRI--is that an option for you on your switch? -----Original

Hi everyone, We have been working with Asterisk for a while now and would really like to expand its capabilities by fully integrating it with our Definity g3si and are wondering about other peoples experiences with similar setups. Thus far we have only been able to achieve a partial 1 way integration, but ultimately would love to route inbound voip calls to the asterisk out of the lucent.

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 successfully integrated * to a Definity G3SI PBX via PRI. On calls from the * box to a Definity display telephone a umlaut over an I appears at the beginning of the caller name on the Definity display. Anyone seen this before? Thanks... Bud

Hi Everyone, So I have been further exploring the integration of our asterisk server and our lucent definity g3si system. I took the suggestion of setting up an isdn-pri line added the two way tie trunk and the signalling group, but can't seem to get the PRI signalling working on the asterisk correctly. I've set pri type to network on the lucent, and pri_cpe in zapata on the asterisk, but

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

Hi, I saw your posting on Hipath and Asterisk.I have some doubts on the same.it would be really nice of you if you can help me out.My Doubt is as follows Currently I am using Hipath HG1500 V3.0 with Opticlient4.0. But i am not satisfied with the performance of Opticlient. I wanted to use SJPhone. Regarding this i had a talk with Seimens guys out here but they talk something ilogical. They told

This is my first post, so please feel free to direct me to another list if needed. I am in the early stages of researching Asterisk. I administer a small Avaya Definity G3 switch (~400 users). Can anyone point my to resources/documents/actual implementation notes of using Asterisk's voicemail with an Avaya Definity switch? Many thanks, Brian Hudson --------------

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

Judging from log analysis, there has been in the past some interest in the Asterisk example files that I have made available from my experimental platform (home office) so I've put my newest versions up on my site. Feel free to dig through these and cut pieces up for your own use, or just marvel at the bad logic flows. :) http://www.loligo.com/asterisk/current/ JT

Does anyone know how to monitor * to see if they are receiving timing slips on a span connected to a T100P card? I am seeing b-channel restarts quite often and also getting "No D-channels available" warnings from time to time. Yesterday I had all the b-channels crash during a MeetMe Conference. Not good! This PRI is connected to an Avaya Definity PBX that is onsite and located in the

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

If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115

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????