similar to: Connecting Asterisk and Avaya Definity By E1. Incoming work, but not outgoing

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

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

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

You should run the 4ESS protocol ("a" on the Definity command "change DS1 board#) as the Definity may not send Display Name for the NI-2 protocol (settings "b" or "d") Remember to make the Asterisk zapata settings consistent with the Definity. On the Definity trunk group form Page 1, change "Codeset to send Display" to "0 (zero)", also make

I've successfully integrated * to a Definity G3SI PBX via PRI. On calls from the * box to a Definity display telephone a umlaut over an I appears at the beginning of the caller name on the Definity display. Anyone seen this before? Thanks... Bud

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 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

Hello, We want to make H323 calls between asterisk and avaya(lucent) pbx. We create node-name,H.323 signaling group,trunk, but we can not make H.323 calls to asterisk. Also no warnings exist in debug. Instead of giving the IP of Asterisk ,i give my computer's IP and run SJPhone ith H.323 GUI. In this time, connection is established. SJPhone accepts H323 calls but Asterisk does not. Do

Hi there I remember to ask this question in the past but now I have thought of something little bit difference. While I understand that asterisk dialplan accept the call to be answered[ Answer() ] in the dialplan, I wanna know if this is possible; i. A call on legacy PBX, extension to extension is made. ii. On call bridging, the legacy PBX initiate a third bridging to the recording system via an

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, 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.

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

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

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,

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

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 --

Hi. After a small war with "underfined sybol" error and conflicts between h323 and oh323 I successfully install h323 channel. Now, I can connect from Netmeeting to SIP and ZAP channels, but I can't here anything. When I call at phone, and try to speak, on another end of line man said, that my voice very low. Microphone volume is maximum... Is there some parameters like rxgain,

Hi, I have two t100p cards installed. One card connected to the PSTN, the other card is hooked to the Definity PBX with a T1 cross over cable. The card connected to the PSTN works fine. The problem is with the card hooked up to the PBX. The light behind the card is green, but when I dial from this card I get an error. Its says it could start the D-Channel. Could someone help me out with this

I'm not a big fan of Line Side for Asterisk applications.... the individual channels have to be reserved (reminds me of the "old" days in telephony) and I don't like that... rather make it a tie trunk and use ARS/AAR routing in the Definity... much more general application. This also makes routing on the Asterisk a lot simpler. Easy to make the changes.... will advise if