similar to: Definity G3R and MWI

We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the remote facilities and try to call the G3R's voice mail system(Audix), DTMF is not recognized unless

Hi, I'm a lab manager / supervisor at our labs. We've had Asterisk in use for over a year directly hooked to the PSTN - a no brainer for configuration (although I had to fix some AT&T specific things in libpri). Right now I have two big challenges. One is to hook our box up lineside to a Lucent Definity G3R. Avaya is chasing this from their end, but we had the highest level of

Hey everybody, I'm looking for some information that I'm not finding by searching the 2004 Asterisk archives. I'm currently playing with a Digium T100P card and 2 Grandstream phones, things are working well. I wanted to move on to linking our Definity G3R Rev 8.2 to the T100P. Everything that I've read so far shows that you need a TN464 to accomplish this. We have a TN767E

Hi All, We have a Avia Difinity G3R to a Avia EPN connected through 2 dedicated T1 dsics, can I use an Asterisk solution to replace the T1's? If so can anyone give me feedback on what components I should be looking into to help me get started? Thanks in advance, Steve.

Is anybody currently running a Definity G3 connected to Asterisk via PRI and a TN464? I've upgraded today and it would appear that caller id info coming across the PRI from the Definity is no longer showing up on the SIP phones. Caller id info from the SIP phones to the Definity is working fine. I'm going to be testing previous versions of Zaptel and Libpri to find the culprit and

I?ve got an interesting problem and am wondering if anyone can shed light ? I am running Asterisk on RHEL Server release 5.2 connecting to an Avaya Definity G3R via a Digium TE220. Asterisk 1.4.20 Zaptel 1.4.4 Libpri 1.4.4 MySQL 5.0.45 Festival Speech Synthesis System: 1.95 We have about 4200 accounts in a MySQL db. Asterisk retrieves the user information from the database, festival tts says

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.

I am thinking of going with a Dell PowerEdge 1950 ||| for a new CentOS/Asterisk set up. It will have dual 2.33GHz processors, 16GB memory, two 500GB hard drives (presumably mirrored). I also plan to get a Digium TE220B to go with it. (a non-dell server is not an option, but I am wondering if there is a better one to consider) The system will be a voice mail repository for 4-6,000 students.

Yes, I can make a call on that extension from other definity phone, if you mean it. -----Original Message----- From: Ken Godee [mailto:ken@perfect-image.com] Sent: 30 ÉÀÌÑ 2004 Ç. 19:14 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing Roman Bessyadovskii wrote: > Hi All. > > I connect

We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA

Calling all Definity admins, Got a few questions about Definity -> Asterisk interoperability. 1) What are the options for integration? Can I hand off extensions from the Definity and vice versa? 2) Anybody have any working configs they would like to post? I've found and read the legacy integration on the wiki about the two systems. I've also googled and found a few threads that were

Has anyone tried to replace Witness or Nice recorder with asterisk. I saw a nice article on voip-info.org on how to replace voicemail server for Avaya Definity with asterisk. The idea behind is to record not only the external channels but also extension to extension (three way calling for which the third leg is asterisk PRI will do) Any suggestion will highly help Sam -------------- next part

Before I go and try to write something myself, I'm curious if anyone has a script that they're using for setting and clearing the MWI on a legacy PBX. I need to pick up a Zap channel and dial #63XXX to set the MWI, or #64XXX to clear it, where XXX is the extension number. One complication is that I've got a couple dozen extensions to handle the MWI for, and only four channels to work

I am hoping someone could shed some light and point me in the right direction? I'm attempting to get callerid work between an Avaya Definity PBX and Asterisk via TE110P connected via T1/PRI Crossover PRI. From the Definity side I've searched endlessly and came with an example which we modeled as close as we can, but still no luck. While doing PRI intense debug span 1 in I see a couple

Hi All. I connect asterisk and definity by manual at www.voip-info.org/tiki-index.php?page=Asterisk%20Avaya. (I just only have E1, not T1 card). I see, that card work (in definity trunk status, and at asterisk == D-Channel on span 1 up -- B-channel 1 successfully restarted on span 1 -- B-channel 2 successfully restarted on span 1 -- B-channel 3 successfully restarted on span 1

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 --------------

Hi all, I'm new to the list, so I apologize in advance if I'm beating a dead horse by asking this, but I read somewhere that asterisk 1.4 has MWI working for Avaya and their rather troublesome SIP firmware. Can anyone verify this before I go changing phone systems around? Thanks Derek -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

We have recently pulled an ancient Fujitsu-branded Centigram voicemail system out of production use and replaced it with an Asterisk box, which is now serving as our enterprise voicemail system and automated attendant. The Asterisk system is connected to a Fujitsu F9600 PBX and uses the 1.6.1.x SMDI module to communicate between the two systems. Save for one issue, the replacement of the old

Hey everybody, I have a situation where we have 2 Asterisk (CVS as of 08/25/2005) connected via IAX. On the corporate side, we have 1 TE110P connecting to a Definity G3R and it's connecting to a TN464F card, giving a 23 channel connection. I have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes. One the remote office side, they a Adit 600 channel bank for 10 outside