similar to: Definity PBX with a T100P & TN767E

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.

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

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

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

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'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

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,

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 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

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

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 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.

If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115

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

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

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

Hello all, I have two questions about anova (one is probably VERY basic...) 1 - when one asks for a summary of a trend surface created with surf.ls, he/she gets: > summary(g3r) Analysis of Variance Table Model: surf.ls(np = 3, x = gradiente$east, y = gradiente$north, z = gradiente$num1) Sum Sq Df Mean Sq F value Pr(>F) Regression 215.7182 9 23.968693976

The problem I was experiencing is still occurring, and it is getting worse. There are several names that Festival "gets stuck on". I don't know if it is a Festival problem or an Asterisk problem. The scenario, a call comes in goes through the dialplan (shown below in original message), and either reads the value ${FULLNAME}, but then hangs. It doesn't go to the next command

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.