similar to: Restart when convenient

I've been trying to setup AddQueueMember() as a replacement for the deprecated AgentCallbackLogin(), but I get _tree_ Queue()'s. Massaged extensions.conf (can provide the original if need be): ----- s n i p ----- [default] include => agent-loginout include => local ; ---------- [agent-loginout] exten => _100.,n,Macro(queue-addremove,I${EXTEN:3},dispatch,10)

Everybody, I have a fall though context that, among other things, tests to see if someone it trying to pick up a non-existent parked call (Defined from 90 to 99). I have the following: [not-in-service] exten => _X.,1,Wait(1) exten => _X.,n,ResetCDR() ; ************************************************** ; Check to see if the mis-dialed number was a parking ; slot. If so, jump to the

Another question, Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our analog lines to a PRI, I thought it would be simple stuff moving the EC to the PRI. Changed signaling, made sure that channel 24 wasn't being ECd and everything came up. But, I was getting complaints of random echo on the PRI. Local echo. Also, we weren?t able to do any kind of modem dial-outs (Adit

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Hi, Can someone walk me through compiling and loading the Zaptel 1.2.10 driver for Mandriva 2006 kernel 2.6.12-12? When I compile and attempt a modprobe I get "module zaptel not found" Thanks Julian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061109/53aed2ff/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On the Python Tutor mailing list Kent Johnson uses a script to find the top posters for the year. If this or something like it has been posted, sorry for the noise; 2008 ==== Steve Totaro 796 Tzafrir Cohen 749 Tilghman Lesher 496 Alex Balashov 354 Olivier 334 Philipp Kempgen 251 Gordon Henderson 242 Atis Lezdins 239 Jay R. Ashworth 230 Doug Lytle 207

Crikey. I can't get this to work! Allegedly, you can press 0 while in the voicemail greeting and be dropped to the 'o' extension. For some reason, I can't get it to work. The 'docs' aren't clear about what context the o extension should be in. The voip wiki says "the context for the voicemail box that we're looking for in the dialplan for the jump to the

Hi List; What configuration needed to let my FXS send and receive FAX? Regards Bilal

I am so happy about learning how to read in multiple Excel files, that I have to try and make another improvement. I know what I have been doing is clumsy, but it works. Hopefully, someone can suggest a more elegant solution. As a novice, I have been using MS-Word and mail merge to write my code. I start with about 2 pages of code, and end up with 2,220 merged pages that I copy and paste into R.

I have the following in my dialplan, counts the number of loops and when it hits greater then 5, exit. It works, but errors initially with, "syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or tolken; Input: +1". Could somebody tell me why? Thanks: ; **************************************** ; Setup a varriable to count the number of ; times the message has been

I am upgrading a box from 1.4 to 1.6 and my console/dsp stopped working. I am getting a SIP/401 Unauthorized error and then a SIP/404 error. I changed nothing in the configs. Is there a particular parameter needed for 1.6 that 1.4 did not care about? If I drop back to 1.4 it starts working again. Thanks Jerry

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

I have a customer who named their endpoint to include a space (example, 1003 a) >From the CLI, I want to hangup a channel on this endpoint >From core show channels concise, I see the channel name includes the space PJSIP/1003 a-00000002 I realize the space is interpreted as an argument separator, so my first attempt below doesn't work. I have tried the following and all fail. hangup

Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.

Anybody having issues with ztdummy under the current 2.6 RC7? I get the following errors when trying to modprobe ztdummy: "Unable to register zaptel rtc driver" Doing a Google on the error shows reference to a message from 2004 that said you might not have RTC compiled into the kernel. Checking via: cd /usr/src/linux-2.6.13-rc7 grep -i rtc .config shows: CONFIG_APM_RTC_IS_GMT=y

This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? -- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in new stack -- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in new stack Bart

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?