similar to: Paging using multiple sound cards/channels

Why *wouldn't* I bother? Using $5-$10 sound cards would be a much cheaper, professional, and more permanent solution than having a couple hacked phones laying near the PA system. Just trying to keep the setup clutter and problem free, as well as as cheap as possible. I'm not a developer, but I don't think it'd be too much of a pain to program the console channel driver to provide

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi there, bristuff comes with these two applications - and too little info to understand what they are for. Anyone has a clue and is willing to share it? Thanks, Philipp -= Info about application 'Autoanswer' =- [Synopsis]: Autoanswer a call [Description]: Autoanswer(exten):Used to autoanswer a call for an extension. -= Info about application 'AutoanswerLogin' =-

Ladies, Gentlemen We are experiencing an unusual problem in our asterisk 1.4.34.. We are attempting to determine if channels are in use before paging to them. This works correctly, as in it pages the phone.. however, we see the error message below on the console... after googling, we discovered limited information regarding the issue... -- Executing [NPANXX7298 at from-pstn:1]

I've got one that I haven't been able to solve. Hopefully someone else has had this issue. I'm using the paging script in free pbx, which appears to: Send a sipheader autoanswer, Create a conferece Add the phone to the conference But if the user hits the page extension, all the phones auto answer, and if they hang-up before the phones join the conference I end up with dozens of

I'm afraid I already know the answer because I've done a lot of searching, but does anyone know of a softphone that supports a central phone book and paging (like the sip autoanswer option of some hardphones) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 david at safedatausa.com

My teeth are on edge after this one. A couple of perfectly good hours of my life, and I still don't know what's going on. . . . The extensions.conf.sample that comes with the current SVN trunk has this line, in an example that shows how to use ChanIsAvail: exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) I couldn't get this to work unless I surrounded the

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Hello As far as ive understood, you can just write Exten => s,n,GotoIf([${AVAILSTATUS} = 1]?autoanswer:fail) ${AVAILSTATUS} would return 1, and "${AVAILSTATUS}" would return "1" Jon -----Oprindelig meddelelse----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne af Brian Capouch Sendt: 27. juni 2006 09:10 Til:

I have been testing * with some Cisco 7912G's, in hope to trash our Nortel system. One feature our Nortel system has that I will need to fiqure out on the * system is paging. Is it possible to page a group of phones (all phones) with announcements? We are a k-12 school and we use our current phone system to make announcements on the phones monitor speaker. Any direction I can be pointed in

Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I

My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the "music on hold" works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag

Hi Thank you for your work. We are trying to use the speex SW with a C6000 TI DSP. I am working with an HW codec that does AGC by HW inside. Will doing the AGC before the preprocessor make a problem? Does VAD work well in a noisy eviroment? In SW there are some remarks on things to be fixed, can I count on SW as working or are some part still to be improved. I also tried to understand SW but I

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

I guess this email is sent to any subscriber. As far as I am concerned, I have never sent anything, because until today I did not have any time to send responses or requests for help, etc. Please do consider the content of your email which can sometimes be ambiguous. Regards Fran?ois T. SAP GLOBAL IT FRANCE SAP Internal IT Support T +33 1 55 30 23 57 (internal 2357) M +33 6 03 53 03 95

Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display "Error Verifying Config Info" in the Status messages and will not process the

Man, I really feel stupid, but after banging my head on a brick wall for several hours ... I need help! I've gotten the BLF to work on my shiny new GXP2010. The GXP is exten 5707, and I've got an xlite on 5608. When I make a call from an outside line, I dial SIP/5608. The little blinky light on the GXP that's monitoring 5608 goes, well, "blink blink". :) I then press

Asterisk 1.4 r181990 given the dialplan snippet below, can anyone tell me why the h exten is not being run ? ============================================================================ console output: [Mar 29 10:33:49] -- Executing [s at questionnaire-menu:1] Set("Zap/1-1", "TIMEOUT(digit)=3") in new stack [Mar 29 10:33:49] -- Digit timeout set to 3 [Mar 29

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