similar to: Double-ring tone

I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? Thanks in Advance, Vic

I'm trying to build a test suite so that I can run "calls" through and verify the call results. I've made a cross over cable and linked my 2 ISDN30 ports together. So now I can dial out on span 1 , and to receive the call on span 2. in the context for span 2, I have the following: <snip> ; #1 "answer" a call and play music 000XXX : ring for a random period,

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 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 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 want to be able to allow a caller to dial a ddi, system to verify identity etc (this is all done) I then want them to sit listening to music, until an event happens. When this (external) event happens, I want to play a certain file to the caller, using playback (so that they have ff / rw etc), and when finished, go back to the music. 1) I thought of redirecting to an extension that played the

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:

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

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

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

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);

Just for some information really : How many of you use multiple sip lines on a phone ?. I'm sitting here looking at my 7960, with it's 6 lines. I've every only used one line, and I was wondering if I was a weirdo ;) The only time I've ever found a use was when I had two systems (production and test) and it caused so much grief (could have been asterisk or cisco) I simply use a

Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml

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

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

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

Can I run two instances of asterisk sharing a single te412p ? I want to be able to have several asterisk servers (for testing various scenarios) running on one server. I was wondering if these asterisk processes could share a zaptel/dahdi card nicely. Julian

Anyone got any subjective (!) views on the merits of these two ranges , using asterisk 1.4 ? I need to supply approx 30 handsets to a new client, with the senior managers (6) having some slightly more "managerial" phones than the base phones which will be used for one line only. TIA Julian -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was wanting to use the new MuxMon application to record calls - it seems to be a "nicer" way of recording than the Monitor application. However, there is a slight issue with agents - we use the recordcalls option in agents.conf to record inbound agent calls - and I believe from looking at the source code that is uses the monitor application. Is there any way to get chan_agent to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 This is slowly driving me nuts! I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk 1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls I get a double ring tone (UK style + US style). I also have a DECT phone on a Sipura SPA-3000 configured with UK tones. This gives me a double ring of UK + UK, so this