similar to: Definity Asterisk Caller ID Issue

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

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

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

Can a temporary message in Asterisk voicemail be de-activated so that the "regular" unavailable and busy messages are played. I have several users who are stuck with the temporary message. Thanks Mark

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

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

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

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, Michael Felt wrote on Mon, Aug 20, 2018 at 11:28:26AM +0200: > On 20/08/2018 10:33, Michael Felt wrote: >> On 17/08/2018 17:15, Ingo Schwarze wrote: >>> Darren Tucker wrote on Fri, Aug 17, 2018 at 07:16:03AM -0700: >>>> On 13 August 2018 at 15:06, Val Baranov <val.baranov at duke.edu> wrote: >>>>> test_utf8: ........................

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, 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

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,

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

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

Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says "Presenation allowed of network provided number" which leads me to believe Asterisk thinks it should not be displaying it. Can anyone

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