similar to: Asterisk to Asterisk SIP Trunk and CallerID

Hello! Is there any easy way to use the caller ID "display info" (CALLERID(name) in Asterisk) in dialplan just as we could use the number in: exten => _X./67803287, 1, <action> I have a SIP GSM device, and when a call comes in, it passes me the caller ID like so: -- Sip message Header: From: "67803287" <sip:gsm@192.168.10.1>;tag=... -- Asterisk variables:

Been working fine for us so far. -----Original Message----- From: Andrew Joakimsen [mailto:joakimsen@gmail.com] Sent: Wednesday, December 06, 2006 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI across multiple servers How well would NFS work in this situation? On 12/6/06, Porier, Jeremy M. < jporier@ccu.edu> wrote: We are about to

Looks like we will be forced to make a move on our voicemail system as Nortel has declared Meridian Mail an end of life product. Frustrating thing is that it would seem their only reason for it is so they can force our hand to move to Call Pilot. Is there any documentation and feedback out there of how people have used Asterisk as a voicemail system for a legacy PBX? We've got about

hi all, I'm trying to send text messages to Snom 300 to show the credit remaining during the call... Sending a MESSAGE directly to the phone via udp i'm able to update the text on the display... but not during the conversation. I read about AOC, but i can't find any documentation about Asterisk + SIP + AOC Have you any experience, docs or workaround to suggest? Thx Ale

Outbound calls on my Telus PRI aren't taking the Name portion of the callerID. I've looked at the logs, and it is being set (see below), but the PRI debug output doesn't show the name being sent anywhere. As a result, received calls always display from Unknown (or just the number). Is there some config that I've missed somewhere? I'm running NI-1 (Telus says NI-2 doesn't

I'm looking to send calls to a phone with no callerid data whatsoever shown on the Polycom as far as missed call. The specific application for this is that I have a 50 phone install with some being used for paging. Paging works perfectly, but the problem is that for every page there is a "missed call" shown on the screen. I have access to the Polycom phone.cfg file, and

I have been unable to get callerid name passed from Cisco Callmanager over a SIP trunk to Asterisk. Only the number is displayed. Has anyone been successful getting callerid name?

Hello all, Recently I just instaled asterisk-1.2.14, zaptel-1.2.12, libpri-1.2.4 and Freepbx v.2.2.0. My zapata.conf look like this, (Pasted bellow) The problem is that the asterisk never send the callerID to the phones. I just take a look to the cdr database an there is no callerid too. I do not know why the calledID is not receibed. All this FXO ports are conected to a mobile lines and if I

I have 2 issues which I need to resolve on our production Asterisk server: We are currently using Polycom IP600 VOIP phones for our office which are capable of handling 2 calls per SIP registration. What we're finding is when staff are on the phone, Asterisk will pass them a second call which will show up on their display, and an audible beep is heard over the phone (regular call waiting). I

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all

I am thinking about a change to our company's phone "layout" and would like to get comments from people who have done something similar. Currently, we have 3 locations - each with their own Asterisk PBX. The corporate office has a PRI. Each remote location has a SIP provider for 5 channels of SIP going to their own PBX. Interoffice calls use the PSTN. Most inbound calls come to

Hi, I have 10 DID numbers. Calls coming from the PSTN network are routed correctly to the SIP users based on the number that was called. But when sip users call the PSTN network, the CallerID should be set to correspondent with their DID number. At the moment I can set the CallerID to a global number, but I have no idea how to check who's making the call. All sip users start in the context

I put my voicemail groups into different contexts so that I can use Dial by name and escape. I had set ext 500 as exten => 500,1,VoiceMailMain(${CALLERID(number)}@default|s) but now that the contexts are different. this does not work #1 how do I have everyone use an ext to get the voicemail regardless of context. #2 can I get the mail buttons to work on my polycom 501s and swissphones #3

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating

Hello list. I have a question about how to read the incoming calls' callerid on an FXO interface of a TDM 400 analog card; (it's one of those RED modules). Now -may this is the complexity adding step..- I have a GSM gateway attached to this FXO thing; incoming calls are processed as they should. But both when peeking on the CLI, as well as in the phone display I do not see the caller id.

We have the caller id with name option enabled with our provider, however, our polycom 501 phones will only display the number of the incoming call. Is there a way to see the callerid name from the cli when the call is coming in (like a print in the dial plan)? I'm not sure if the problem is with asterisk or our phones. I did turn on the calleridpres option in zapata, but I'm unsure what

Ok, I must be really stupid here - I'm playing with ael and svn trunk. given the following in ael: context isdn10 { 444601 => { Answer(); NoOp(${CALLERIDNUM}); Hangup(); }; }; isdn10 is the incoming isdn context. why do I get this on the console: -- Accepting call from '01702xxxxxx' to 'yyyyyy' on

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

Howdy, Working on some niche requests from one of my hotel clients. asterisk 1.4.20-1 on CentOS, Polycom 501s. The first request is for the Polycom's screen to show the CID of the inbound caller when a call pick is executed, so the picker knows if the call is internal or external. I have already "worked around" this issue by using ALERT info to give seperate ring tones for