similar to: PRI CID signalling not working?

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

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 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

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

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

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

Hello everyone - first time poster, long time lurker. (sounds like a radio morning program, I know). I'm attempting to get my InterTel Axxess (w/v9.0 software) to play nice with my Asterisk implementation. Asterisk 1.2.6 is running on a Fedora Core 4 x86 box. I've tried getting the Axxess to talk SIP to Asterisk, but InterTel's SIP implementation is, well-let's say, incomplete.

I have a handful of Linksys PAP2-NA's all talking nicely to Asterisk using standard telephones. I've been running them for the better part of this year. No complaints whatsoever. We chose the PAP2-NA's mainly due to cost and especially the ease of provisioning. In an effort to inexpensively bridge our office PBX (InterTel Axxess) to our VoIP network, we've opted to connect

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

I have asterisk FXO module on a TDM400P hooked to an Intertel Single Line Card. I can place Intertel intercom calls to Asterisk (both SIP and analog phones) and the reverse, but transfering calls doesn't work. Here what the transfer looks like: Intertel FXS > Transfer > Asterisk FXO > Asterisk FXS > any extension When the call is answered Asterisk hangs up. Intertel

Is it possible to act as a middle man on a T1 line? My installation currently has an aging Inter-Tel Axxess box with a T1 coming in (16 in, 8 out). Rather than adding and replacing phones and cards as they die, I would like to slowly migrate to a asterisk SIP installation. I want to take the incoming T1 line, use any available outgoing lines for outgoing SIP, intercept any incoming lines and

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> | | | | | |

Hello All, Just looking some comments from gurus about this proprietary systems and phones: Inter-Tel Eclipse2 Model name: IP PhonePlus I did not find anything useful or reasonable about their products on their website or even in Internet.... except sales. -- Thanks and regards, Vasyl Rublyov

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

My legacy PBX accepts CID number, but not name. My old PRI vendor never sent the name, so there was never an issue. I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI - asterisk - PRI - Legacy. Any calls from asterisk (sip and iax extensions) which have callerID set, will not connect. The legacy PBX hangs up, but asterisk thinks that it is still ringing. I have added

I have TE405P span connected to a patton/digi 2977 T1/PRI port. Zaptel starts up fine without any errors. Libpri is installed as well. But I get the following when trying to load asterisk ------- Jul 3 19:30:26 ERROR[8192]: Signalling requested is PRI Signalling but line is in Unknown signalling 896 signalling Jul 3 19:30:26 ERROR[8192]: Unable to register channel '25-48' Jul

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

On Wed, Aug 03, 2005 at 11:28:19AM -0500, asterisk-users-request@lists.digium.com wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <gdh@laterooms.com> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span

I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending "reset all channels"

Yep, another list posting on this topic :) All the messages I've read on this are from people experiencing these errors in quiet times - I get them as soon as I plug a port on our TE410P to an Inter-Tel AXXESS PBX.. and I get them continuously... I'm just sticking an * box in between ISDN30e (we're in the UK so euroisdn) and the PBX.. and whilst the telco ISDN30e side works like