similar to: Avaya Partner Phones to SIP?

Does Asterisk work with Lucent or any other PBX phone systems????

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

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

The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it's a SIP phone or

I'm trying to connect an Asterisk system to an Avaya Partner ACS R6 system. The problem I'm having is that I cannot get the partner system to get CallerID over the T1 modlue. The partner is using the T1 with E & M signalling (which I don't think can be changed), and whatever I tried didn't work. My only option right now is to get FXS ports on the Avaya side plugged into the

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 --

So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23] WARNING[6227]:

Hi, I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using chan_ooh323 from asterisk-addons. I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station (phone) and vice versa. I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN. However I face problems when I make DID calls from the PSTN. The DID calls are made through

I have added H.323 support for ASTERISK PBX using the OpenH323 library. I thought that since the Avaya IP phones are H.323 phones I will be able to make them register to Asterisk server. Does anyone know if this is possible ? Katerina

Hi I've a softphone account to a AVAYA H.323 system, basically, it has a numeric ID (which is the extension number) and a numeric password. Instead of using the default AVAYA softphone (H.323), can I make asterisk as a H.323 client and login to the AVAYA system via any one of its h323 modules? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just connected an Asterisk server to an Avaya Pbx using the instructions at: http://www.voip-info.org/wiki/index.php?page=Asterisk +Avaya Everything seems to be working as I can send and receive calls. The only detail I am having a problem with is that when an extension on the Avaya calls an Asterisk extension, I only get the name part of the CID, not the number. From Asterisk to the Avaya

Hi List, Ive got a bunch of Avaya IP9640 that we want to convert to SIP and then hook up to Asterisk so we can dump this overpriced Avaya system. Ive got ahold of the SIP firmware, but I cannot find anything on how to convert the phone itself to SIP, when I go into setup mode it wants a "command" which im guessing is the program code like the rest of the Avaya systems works. Has anyone

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

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

Ok, to review, I'm trying to get Avaya 9608G to come up in a pure Asterisk environment-- no Avaya SBC or gateway or any other Avaya gear in sight. I have the phone working to the point where it boots up properly, then displays a Username and Password prompt, and says its extension is 123 and the time is 4:57p, which is wrong. But please don't tell me the only way to program up each

I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.

I attempted to update an Avaya 4602 phone to the latest SIP firmware and now the phone stops at the bootloader. It keeps requesting an appsip.ebin file from my HTTP server and is no longer checking my TFTP server for update files. Since no appsip.ebin file was included in the firmware update provided by Avaya, I have no idea where I am supposed to get this file. And also since it is no

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

n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, --