similar to: CVS HEAD 08-11-2005 & Definity G3R

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,

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

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 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'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

Hi there, So I've finally got our Definity and * box talking back and forth, but can't figure out how get callerid sent from the Definity to *. Has anyone had any success with this? I've tried every combination of zapata.conf variables pertaining to callerid with the same results: Accepting call from '' to 'xxxx' on channel 0/1, span 1. On the SIP phones I receive the

I could only get *ANI*DNIS* working one way and that was setting my signalling type on the Asterisk side to 'featd' The Definity won't send *ANI*DNIS* information back to the asterisk as far as I can tell. Other than that, I've been running it with wink/wink E&M for a while now. TN464 circuit pack on the definity side, te110p on the asterisk side. Just a crossover cable

Thanks for you comments. I have the one port card now. I plan on purchasing the TDM400. My only question is whether or not the Dell optiplex has pci 2.1 (I think) I have one system running with the single port card. Today I received 2 sipura 2000 modules. Configured the first one tonight... Works great. Pat -----Original Message----- From: asterisk-users-bounces@lists.digium.com

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

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

AS a proof of concept experiment, I want to try and integrate Asterisk with my Lucent Definity G3 switch. I don?t have an available T1 port on the G3 but I can round up 4 analog ports off the G3. What I thought I could do is create a ?Hunt group? on the G3. Let?s say I configure 5610 ? 5613 as a hunt group. I know I would need a 4 port FXO card to install in the Asterisk server. Two configuration

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

Anyone having problems compiling the current cvs head this morning? New cvs checkout on RH9, followed by appropriate make clean and make install. System was running cvs head from Nov 23 with TDM card, PRI, SIP phones on local wire, and IAX. Appears zaptel and libpri compiled correctly, however the first attempt in the asterisk src directory yielded: gcc -pipe -Wall -Wstrict-prototypes

Has anybody has issues with the new Native MOH and IAX trunking when placing a call on hold? My scenario, Call is placed on a Definity G3 via PRI to Asterisk. Gets trunked over to another Asterisk system via IAX2. Call is answered by operator and placed on hold. At that point, audio is very broken and feedback pulsing is heard. Bad enough that the caller hangs up. If the call is

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 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

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

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

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