similar to: Integration to TAPIT/Call tracking software

Hello, I have a customer who has their front door integrated to their current phone system. If someone presses the buzzer, the secretaries phone will ring, and she can talk to the person at the door. By pressing ** she can release the door. Anyone have any sort of integration like this. Are there IP devices anyone is using? They have a pizza box server as their asterisk server with

Hello, I am wondering if anyone is using the Nortel 2001 2002 or 2004 phones on their asterisk implementation. My local dealer says they are not compatible with any open source implementations. Is there a phone compatibility list somewhere? Cheers Cian

Hello, I ordered the Devel lite kit, and installed it. I am just trying to get the FXO port to work, and am having trouble. To load the card I do the following. modprobe wcfxs modprobe wcfxo ztcfg -vv asterisk -vc My /var/log/asterisk/messages show Nov 3 11:03:39 WARNING[3317]: zt hook failed: Device or resource busy Here is my /etc/zaptel.conf fxoks=1 fxsks=4 loadzone=us defaultzone=us

Buenas noches; Traigo una pregunta que supongo que alguno ya la tendrá resuelta, porque se me hace difícil entender algo que presupongo fácil. Quiero, según los valores de 3 o 4 variables numéricas, convenientemente escaladas, conseguir gamas de colores. Supongamos las variables numéricas: X, Y, Z; a cada variable le correspondería un color; pongamos que X = C (cian), Y = M (magenta) y Z = Y

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

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.

Dear colleagues: This summer, the ASA will award a new prize for statistical software. This note is to encourage you to apply or nudge someone you know to apply. The award is for software that contributes to better use of computing in statistical research and applications, written while the applicant was a student. There is a $1,000 prize plus the opportunity to give an invited talk at the

Some changes to the implementation of S4 classes in the methods package will be committed to the r-devel version, probably later today. They should be largely back-compatible (exception noted below) but they are fairly extensive, so this is an advance notice. The changes form the first of a two-stage change aimed at making the implementation more consistent with R semantics, in particular with

Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 22, 2006 8:26 PM To:

Hello dear R-help mailing list. Looks like the same issue in Russian: library(RCurl) library(XML) u = " http://www.cian.ru/cat.php?deal_type=2&obl_id=1&room1=1" a = getURL(u) a # Here - the Russian is fine. a2 <- htmlParse(a) a2 # Here it is a mess... None of these seem to fix it: htmlParse(a, encoding = "windows-1251") htmlParse(a, encoding =

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

I've read the wiki and other resources on how to connect Vonage / Voicepulse and all these other services to Asterisk... We are attempting a connection to a Lucent iMerge. Lucent has told us that it won't work - but we feel confident that it will. Has anyone worked with the Lucent iMerge - or would be willing to help lend a hand? It is capable of H323 / MGCP. Even if I could make the

Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

Hi all, I am new to asterisk, but experienced with Linux and Avaya/Lucent G3R hardware/software. We currently have a large inventory of spare Lucent 64XX phone sets. I'm considering setting up an asterisk system for a remote office, and would like to use the Lucent sets because they are cheap, and familiar to my end users. Does anyone know of a card that can drive these sets? I think

I've posted this before but before it was just wishful thinking. But now it has become a real need. Our boss has a 5 year grudge against Lucent. They had a Lucent here that would simply stop processing calls. No alarms. No nothing. I'm sure that in 5 years, new software has been released. I've got another guy running several AS5400HPX's telling me that you can only do about

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Troy Settle > Sent: Wednesday, September 14, 2005 7:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not > enoughlinesavailable for Asterisk implemetation)

Ok, from what I can see _NO_ passive ISDN cards will work with Asterisk on freebsd, is this correct & is it likely to change soon? Secondly, if this is likely to be the way for a while, what is the lease expensive card that will work with FreeBSD? Also, can I use DID (Direct Inward Dialling) on FreeBSD? Thanks for all your help to date. Regards, Cian Hughes

I am running wine 20050725. I used winetools to get an assortment of software loaded and with a bit of tweeking it worked. I then ran the install CD for Broderbund "The Print Shop". It loaded OK and after copying a few dll's from my windows box, it came up and ran. My problem is with printing. After reading through the users guide on printing and poking around in the archives

I have a problem where an Asterisk server is sending a premature DRQ... Not sure why.. Here's the setup - Asterisk using inAccess networks H323 replacement channel driver Connecting to a Lucent iMerge... The call connects fine - I get the out of the box greeting - but after exactly one Minute - the call terminates. I have had this problem on multiple different Asterisk configs... I'm