similar to: CONNECTEDLINE and panasonic 500

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

Does anyone on this list know of any Linux based apps that will work with Dialogic or Brooktrout that provides voice-mail/Autoattendant only?? It seems that Panasonic, Avaya, and Mitel all use Unix/linux based OS on their firmware for their proprietary voice mails. My wish list would be; A software that provides all of the drivers for a dialogic or brooktrout board Voice Mail Messages in WAV

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or cisco gateway) I don't see this. Could you tell me is it possible? Thank you!

Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice ----------------> Asterisk ------------------->Bob ------- INVITE

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group 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 Guys, I'm looking for a light on the next problem integrating an Asterisk server with a Panasonic PBX using a PRI. The follow config has been working for two weeks and works almost fine, except for some little problems... 1- PSTN provides the E1/PRI (EuroISDN) 2- Asterisk receives the PRI 3- Asterisk provides a second PRI to Panasonic 4- Panasonic receives the asterisk PRI People can

We have a setup running Asterisk interconnected to a Panasonic TDA200. The Asterisk server has a two port E1 card, one connected to the phone company and the other to the Panasonic. Everything is running fine and we can send and receive calls from the Panasonic and phone company. We are using MFC/R2 for both links on Asterisk 1.4.4 and Zaptel 1.4.3. The only detail we have is that we cannot

Hi all, I have an existing panasonic analog pbx in use and a asterisk server with digium tdm400p(2 fxs and 2 fxo). channel 1 -> fxs -> telephone channel 2 -> fxs -> telephone channel 3 -> fxo -> extension 15 at panasonic pbx channel 4 -> fxo -> phone line from telco We call in to fxo (channel 4) and enter the ivr which prompt us to enter the extension number. After

Hi, Anybody have an idea how to integrate * with a Panasonic td-1232? We one at the main office, and are installing * in a branch office. We'd like to be able to make calls from * extensions to Panasonic extensions and the other way around. Making outgoing calls from extensions one one side to lines on the other would be nice too. I can put another * machine at the main office, but what is

Dear Pals One customer has a Panasonic PBX KX-T336 with 60 ext. and a E1 (R2) for Trunks working perfectly now, This customer has 10 wireless links to his branches, wireless working great now, no voice at the present MY IDEA : T1Card into the Panasonic (additional to the E1) connected to a T1 Digium Card into Asterisk as far as I know the T1 Card can be configured as E&M to act as

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

Hello! I have asterisk which is connected to avaya definity. I set trunk to overlap. When I call to this trunk (so called tac in avaya) without any number I hear dial tone for some time, any digit I try to dial are ignored by asterisk- still tone, then call is rejected: -- Accepting overlap call from '6401' to '<unspecified>' on channel 0/9, span 3 -- Starting

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

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

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

hi, anybody know how to make a sip trunk between trixbox and panasonic kx tde 100? i've tried, but always failed when calling from trixbox to panasonic. thanks,,, -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130312/1dd5170b/attachment.htm>

Hi List, I have another issue on allowing outgoing calls to PSTN on Asterisk via Avaya Phones, I hope that anyone could help me fix this issue: *When I dial through Avaya phone i just here a "good bye message" reply from asterisk server. And here is the log:* == Starting OOH323/(null)-b7db8aa0 at internal,s,1 failed so falling back to exten 's' == Starting