similar to: a few question about asterisk

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

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 -

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 everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hello - I have (finally) obtained a fair amount of success in connecting my Intertel Axxess PBX to an asterisk box via a T1 PRI. I can place calls from the Intertel side through the T1, out to an IAX2 softphone and the calls get routed correctly and all of the CID information stays intact. However, when I call from the IAX side to an extension which should route back through to the Intertel

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

Greetings all - I'm interested in using an asterisk box to supplement and add VoIP capabilities to our legacy InterTel Axxess PBX. After searching through the list archives and through google, it seems that the best way to go about this is to connect the two systems via a T1. Is this correct? The PBX currently doesn't have any VoIP capabilities, so that's not an option for

Hi all, My office recently purchased an InterTel Axxess system with the IPRC card for VoIP. To our suprise, this card allows the InterTel endpoints and MGCP endpoints to work, but not SIP clients. I was really expecting to get a SIP softphone working with this setup, but that appears to require our vendor to sell us a SIP gateway and licenses at a not yet determined price. With this

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

I've been trying to figure out if it's possible to connect Asterisk to a parent Inter Tel Axxess system through the MGCP protocol. The archives for this list aren't searchable and I'm wondering if anyone has a simple answer... Dustin Moore

Hi all, I want to setup a meetme application in the following maner: One operator is connected to a room. The operator hears and can talk to all the participants, but one participant can only hear/talk to the operator, not others. The operator is using one phone. To be more explicit, this means that every new person etering the room has a one2one conversation with the operator only, and the

Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone system via a SIP trunk using the IPRC card? -- Marvin Horst -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/5dbe435a/attachment.htm

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

Anyone know of a business class sip hard phone that includes a quality display capable of supporting "push" data (maybe Polycom?). Something like... VM: 3 msgs OurStock (1:43pm): 59.5 somewhere on the display that can be updated (pushed) from a server? Rich

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Good Day list, Need assistance determining the best place to read up on whether Asterisk can help me out. I have a situation where I need to do the following <PRI from Telco> ------- <Analog Channel Bank>------------<Proprietary Box> | | | | | | <PRI Port 1 of Digium Quad T1> <PRI Port 2 of Digium Quad T1> | | | | | |

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

I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can successfully make a call from the SIP phone to any other phone (inside or outside), but I can not make any calls to a SIP phone. Attached are the pertinent parts of sip.conf and extensions.conf. The log starts off normal with: Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 2 on Zap/55-1 Mar