similar to: integrate Intertel Axxess with Asterisk

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

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

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.

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

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 want to use extension numbers that begin with the # key in my dialplan, but I can't get my Aastra phone (6731i) to transmit the # key to asterisk. It works fine for the * key. I've tried numerous Local Dial Plan patterns in the aastra web configuration but none of them worked. My current Local dial Plan pattern is "x+#|xx+*|#x+". Any help would be appreciated. -- Marvin

I have been using the D-Link DPH-540 wireless VOIP handset, and I really like this phone. We had tried the UStarcomm phone, but the phone is used in a noisy environment and the volume wasn't loud enough. The "problem" with the D-Link phone is the Li-ion battery needs to be replaced and D-Link doesn't sell a replacement battery and I haven't found any after-market batteries.

I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for "Always" works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to "No Answer" means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a

Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

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

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

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 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

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

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

On Wednesday 03 August 2005 17:33, Jens von B?low wrote: > Gavin, > > >> Any ideas/advice would be warmly received right now! > > You are not going to like my response... Erk :) > The only way I could get this to work (luckily I had 2 identical sites and > was busy with the upgrade to the gen2 card) was to downgrade to zaptel > 1.0.7. Alas no - just moved down to

-----Original Message----- > -this is very true, however, the current version of the Axxess software > (9.0) supports SIP trunking natively on the IPRC. I just got my Axxess > upgraded and am salivating to get * connected to it. Hmm, so 9.0 is out and it supports SIP natively. How did you plan to integrate the 2? -The Axxess will see the * as it would see an IP service provider.

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

Hi, There is a note in the document: *For a password database it?s enough to have only the user and password fields. For a user database, you need to set also uid, gid and preferably also home (see VirtualUsers). (gecos) and (shell) fields are unused by Dovecot.* You can leave empty what you don't need. On Thu, 23 Mar 2023 at 11:09, Horst Simon <horst.simon2 at icloud.com> wrote: >