search for: callernames

Get the new firmware - it's supposed to have changed the callerid display presentation to include name and number. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on

I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe <6092521155>") in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name & Number will show.

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi there, I want to add 4000 Callerids and Callernames to my asterisk-db. (/var/lib/asterisk/astdb) I do not want an external database or an sql-database because I do not want asterisk to depend on external processes. However, when I do 4000 "database put number name" via a shellscript and "asterisk -rx" I only have 600 entries l...

Alright I know how to do basic IVR in *. But what I'm working on trying to do now is a survey. I've found very little things out there on google or the archives for how to do surveys with the * ivr. Here is more or less what I'm trying to accomplish 1. Call comes in Plays Greeting 2. Starts Survey 3. Ask Q1, Record the answer (voice responses) repeat this

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have the 7970 xml config for sip yet? Aaron

I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this

Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know

Hello All, I was wondering. What are the primary advantages to using a PRI over a T1? As I understand it, the PRI terminates very fast, meaning you can do immediate answer and dial... This is very handy on the BRI line I have on the asterisk. Can T1 signalling also do immediate answer, or does it just behave like a channelized pots line and ring as usual? I am trying to determine if I should

Hi, I'm currently working on a setup between Asterisk and VoiceGenie (which is a IVR system). The way my setup is done, is that I have a PRI line coming in my Asterisk server, and then VoiceGenie is connected to Asterisk via SIP, like any other softphone basically. I'm able to receive calls in Asterisk and then link them with VoiceGenie. But one of my issues is that when I get an

Hi all, I'm having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the

[Reposting, as was bounced for non-member, sorry if this is a dupe] Arrangement: { PSTN }--E1--[PBX]--E1--[*]--LAN--[SIP phones] \__[PBX system phones] Normal calls between PBX system phones and SIP phones work, in both directions. The call logs look like (ignore the no answer, it did ring):

Hello guys, We're currently working on asterisk trying to create our own SIP phone, because we need special features. But dunno maybe there's other people who already done that before. Basically, we are a inbound call center. We have serveral customers with different phone numbers, which are redirected to us. When we receive a call coming on a specific phone number, the call gets

Copying to asterisk-users, as it's of use there too. I copied this code years ago from the net, it may have been modified since... This however is only used by managers, as it allows the manager to log a user in and out. For agent logged in/out status: where 8501 is the queue number and 8512 is the agent's extension, and SIP0001 is the agent's device. in extensions.conf