search for: calleridnames

I edited the calleridname.agi patch to only overwrite the name if it is missing. The asteridex option still overwrites the name since it is our master list for known numbers. -- Steven calleridname.agi.patch: --- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006 +++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006 @@ -16,6

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Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

Previously, when I wanted to forward to incoming callerid when I forwarded a call to another number I had to set the callerid on the outgoing call to be that of the incoming number. So today I do this: exten => s,n,Set(CALLERID(name)=${CALLERIDNAME}) because I want the outgoing callerid that I forward to not be the normal callerid of the local extension but I want to forward the incoming

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

I'm having trouble getting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe

Hi, I'm using mysql to define my sipfriends. When authenticating, the calleridname from the calling SIP user (phone) seems getting lost. With "sip debug" I can see in the SIP messages: From: "myName" <sip:4912345@1.2.3.4>;tag=22668125 To: <sip:004954321@1.2.3.4> but I can't find "myName" in any channel variable. Both, ${CALLERID} and

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Adams, Gavin Is there any additional information I could provide to start tracking this down? I was thinking about looking into the various applications source to see how they access the data elements for callerid. I know where the values are pulled

I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says "caller id unavialable". With the spelling done right.... Any ideas on how this

Hey all, We've upgraded our PRI trunk to support what BellSouth calls "enhanced caller id name delivery". The weird part is, I'm only capable of seeing the name portion on incoming calls within voicemail2's email delivery. For example, on an incoming call, asterisk is reporting this: Context from extensions.conf (BS delivers 7-digit DIDs): exten => 9133727,1,Answer

Hi everyone, I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configuration option or something else. When a call from a phone is sent to the * box it has the following sip headers: From: "From Phone" <sip:1001 at 10.0.0.29>;tag=4bf4bb4e11e92476. Remote-Party-ID: "Cloutier"

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Hello All, I have new TDM04B installed and working fine with Asterisk 1.0.5 built on RedHat 9. All is working fine except CallerID that bothers me big time. I have several Panasonic and Sony phones and CallerID works fine with it (when I plug in the line into phone instead into Asterisk I get CallerID) but fails with Asterisk. I am based in California (San Francisco) and my Telco is SBC

HI Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

I have working with a polycom IP500 phone. I like the idea of having each line button on the phone as a separate sip device. If I understand it right, each phone could have three extensions (one for each line.) This would be great since I could then use the dialplan to forward calls to the desired extension. I envision something like this: Extenson 101 - Company-A Extension 102 - Company B

Kevin P. Fleming wrote: > Falck Kenneth wrote: > > Thanks, I was misguided by > > http://www.voip-info.org/wiki-Asterisk+Variables which > didn't mention > > this. > > You are more than welcome to edit the page to make it obvious > to the next reader :-) You're quite right - I added a little note there to warn others now. It's a great Wiki anyway,

Hi all, When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this? -------------- next part -------------- An HTML

I've some code to manipulate incoming Caller-ID - so its suitable for replying to... [sipdef] exten => s,1,NoOp(FWD SIP: "${CALLERIDNAME}" <${CALLERIDNUM}>) ; Alter incoming calles from pulver - add a '87' exten => s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4) exten => s,3,SetCIDName(87${CALLERIDNUM}) exten => s,4,SetCIDNum(87${CALLERIDNUM}) exten