similar to: Re: Definity ISDN PRI

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'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

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.

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

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

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

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

I got the following compile error in R-1.9.0 on NetBSD 1.5: <-- snip --> ... gcc -I../../src/extra/zlib -I../../src/extra/bzip2 -I../../src/extra/pcre -I. -I../../src/include -I../../src/include -I/usr/local/include -DHAVE_CONFIG_H -O2 -mcpu=v8 -c main.c -o main.o main.c: In function `setup_Rmainloop': main.c:463: `CODESET' undeclared (first use in this function)

On 27/03/17 17:06, Tom G. Christensen wrote: > On 20/03/17 14:31, Damien Miller wrote: >> OpenSSH 7.5 has just been released. It will be available from the >> mirrors listed at http://www.openssh.com/ shortly. >> > > I'm seeing an error in the testsuite on Solaris 2.6: > test_utf8: ........................ > regress/unittests/utf8/tests.c:48 test #25

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, 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

Hi, Michael Felt wrote on Mon, Aug 20, 2018 at 05:00:17PM +0200: > ./nl_langinfo > setlocale -> "C" > nl_langinfo -> "ISO8859-1" Thanks, that is helpful. So i think i was wrong and Damien was right. This means that OpenSSH returns truncated messages when non-ASCII bytes occur in them, even when the user requests LC_CTYPE=POSIX. That's not good. While

In case you're ok with not version-controlling files imported via autogen, ... >From 72c829395bb6a4800516d4f535e18af48195585b Mon Sep 17 00:00:00 2001 From: Jim Meyering <meyering at redhat.com> Date: Mon, 3 Aug 2009 11:00:22 -0400 Subject: [PATCH] Remove files imported via autogen.sh. * m4/.gitignore: Ignore these files. * m4/codeset.m4: Remove file. * m4/gettext.m4: Likewise. *

Hi I have done everything richard told to do ECT . below is my trace, anyone can help ? -- DAHDI/i1/09123278669-4 answered DAHDI/i1/88050048-3 -- Native bridging DAHDI/i1/88050048-3 and DAHDI/i1/09123278669-4 PRI Span: 1 Adding facility ie contents to send in FACILITY message: PRI Span: 1 ASN.1 dump PRI Span: 1 Context Specific/C [1 0x01] <A1> Len:11 <0B> PRI Span: 1

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

Ok, so I've made progress on 2BCT (2 B-Channel Transfer). I'm assuming that the debug info below shows that XO doesn't have 2BCT enabled on my line, but if anybody can confirm that'll let me be way more indignant. J -- Native bridging DAHDI/1-1 and DAHDI/3-1 > Protocol Discriminator: Q.931 (8) len=28 > Call Ref: len= 2 (reference 801/0x321) (Terminator) >