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I have a couple of numbers that are diverted to a number that is conected to an isdn30 card, running asterisk 1.4. eg. 123456 => 22334455 654321 => 22334455 What I would like to know is the number of the orginal number dialled (123456 or 654321). I thought that RDNIS was the answer, but it is always coming up blank. When I did a debug on the pri span, I saw the following message

I am playing around with dynamic meetme conferences, and wanted to have one person constantly in the conference, with calls "popping in and out". Is there an option / any way of playing enter / leave sounds to the person who created the conference only, and not the people leaving / joining ? TIA Julian.

I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)" I cannot get the card working in any of the slots.

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

...he call is not passed onto the member, and there is no activity on the cli. Eventually the call times out. If I add SIP/706 instead of Local/706@AgentQ then it all works as expected. Any clues or help ? Many thanks ! Julian. Kevin P. Fleming wrote: > ----- Julian Lyndon-Smith <asterisk@dotr.com> wrote: >> Now, I want to be able to use a device, rather than agents. So I can >> use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device is called. > > This is what the Local channel (chan_local) is for. > >...

Has anyone in the UK got ANI to work on an inbound call ? Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30 Julian ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email

I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on

I've had several users today inform me that whilst they were on a call, the volume kept fading in and out to such an extent that they thought the caller had hung up. I would dismiss this if it were a single person mentioning it, but it isn't .. Has anyone else seen anything like this ? We haven't, and we've been running for nearly a year ... CVS head as of 3 days ago, TE410p

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

...; > > > Today's Topics: > > 1. transparent bridge and proxies (Julian Lyndon-Smith) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 24 Aug 2006 18:31:05 +0100 > From: Julian Lyndon-Smith <asterisk@dotr.com> > Subject: [Bridge] transparent bridge and proxies > To: bridge@lists.osdl.org > Message-ID: <44EDE259.4050009@dotr.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > I want to be able to install a box that is a transparent bridge, but > that i...

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is

...who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline ("asterisk" at "dotr" dot "com") if you want to sell yourselves :) Julian.

Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients...... plz anyone suggest me.... Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/aa4e409c/attachment.htm

We've managed to setup spandsp to receive faxes and email them to the appropriate person. We did all of our testing using Thunderbird, and the attached pdf files worked very well. However, when we went "live", some people complained that the attachments in outlook were named <<subject>>.dat instead (for example) of 123456.pdf Having looked into the mime-construct

When recording inbound agent calls, if the queues use agent members (Agent/6000), we can get the calls recorded as agent-xxxx.yyyy.zzzz.gsm where xxxx is the agent number. However, if the queues use phone members (SIP/6000), the recorded filename is simply yyyy.zzzz.gsm. Is there any way of making the recorded file either agent-xxxx or even sip-xxxx where xxxx is the extension number. I had

Is there anyway to force an autopickup on a cisco 7940 / 60 from the dialplan ? My problem is that I am originating a call from the AMI, with the internal user being called first, and then connecting to external user. However, sometimes the internal user doesn't pick up the phone, so the call is never placed. I need to know the results of the call so I need to be able to either a) get

I experience random crash of machine (full hang, requiring a hard reset) after trying to test run Asterisk 11. The machine is a centos 5.8 32 bits pc with 1G ram. Asterisk is compiled from the source and no other software has been installed Anyone experience similar situation? -------------- next part -------------- An HTML attachment was scrubbed... URL:

If I have the following in the dialplan exten => foo,n,Dial(SIP/1234&Zap/G1c/55443322) and SIP/5432 calls this extension, is it possible to show different callerid numbers to each of the target numbers ? The reason I ask is that if the call is from an internal sip phone, I want to show the internal callerid (5432) to the SIP phone on 1234, and the DDI of the 5432 extension

I'm trying to set a channel variable and make it available to another channel: I thought that if I SetVar(_SomeVariable=SomeValue) or SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in the destination channel. However __SomeVariable, _SomeVariable and SomeVariable are all blank. The scenario: Agents logon to the queue using callbacklogin. From what I can gather

...---------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081129/977731ca/attachment-0001.htm ------------------------------ Message: 15 Date: Sat, 29 Nov 2008 13:19:19 +0000 From: Julian Lyndon-Smith <asterisk at dotr.com> Subject: [asterisk-users] GSM gateways - which one ? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <49314157.4040703 at dotr.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed I've been ask...