similar to: First Post: Any existing AVAYA Switch -> Asterisk Voicemail configs?

Anyone have an H.323 trunk tied between their Avaya and Asterisk box that works? I am having some issues trying to get the two systems to connect. I am using the ooh323 channel to try to make the connection between the two system. I have all my configs if anyone would like to look over them. If I do a trace on Avaya I get a denial event 1191: Network Failure. Thanks! -------------- next part

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

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

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

I am tasked with linking an Avaya Definity switch to an asterisk box using it's IP card that handles H.323. All my googles turn up a lot of results but nothing recent. I am able to find instructions but they are dated from 2005, and often fail halfway through. What is the best way to achieve what I want, which is two way calling between the Avaya switch and Asterisk server using h.323, and

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

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

Does anyone know if avaya voip product is running linux under the hood? Thanks, /glen -- Glen Ford gford@idiom.com

Hi all, I've got an old avaya partner acs <7.0 system here. I'd like to add a simple voip bridge so I can hook up our remote offices. From my research, it would seem the pre-7.0 series doesn't have a t1 port, so if I wanted to do this, I would have to feed the avaya system fxs ports from the asterisk box. Does that sound about right? Has anybody ever done this? Does

Hey everybody, I've just spent the last two hours Googling and searching the Wiki. I'm trying to find if there are any listings of codes for the Avaya Definity G3R, to allow for an Asterisk system to turn on/off a phones MWI that is attached to a G3. We are looking to use an Asterisk system as a voice mail server. I'm not having any luck, anybody have such information? Thanks,

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

Right-- I've seen the Avaya document you cite below. It says "To administer DHCP option 242, make a copy of an existing option 176" but I don't have any example of option 176 or 242 to copy, and don't know what to do to /etc/dhcpd.conf to make it offer option 242. Then there's this long table of parameters to use with (presumably) option 242. I was hoping someone had a

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 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

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

I'd like to start configuring my Avaya 9608G phones for use on Asterisk / FreePBX / PBX-In-a-Flash. I'm using a variety of other phones on my system without major issues. I've read the discussion back in March, May and August of 2016, but unfortunately, my difficulty is much more basic. I think it has to do with DHCP, specifically, what options I'm offering the phone via DHCP.

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

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

I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call