similar to: Re: Paging using multiple sound cards

Does anyone know if its possible to have more than one sound card in an Asterisk box and use each one as a paging zone? How about left and right channels of a single sound card? I'm looking to have 2 paging outputs if possible - I've read about using a Grandstream phone on autoanswer but I'd prefer to have the feeds come directly from the * box and go into a stereo amp feeding 2

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Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

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

The MessageLabs Virus Control Centre discovered a possible virus or unauthorised code (such as a joke program or trojan) in an email sent by you. Please read this whole email carefully. It explains what has happened to your email, which suspected virus has been caught, and what to do if you need help. ------------------------------------------------------------ Some details about 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

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

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 just acquired a copy of "Statistical Models in S", I guess most commonly known as the "white book", and realized to my dismay that most of the code is not directly executable in R, and I was wondering if there was a source discussing the things that are different and what the new ways of calling things are. For instance, the first obstacle was the solder.balance data

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

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

Hi, I use a simple script to query a 2 table's in a MySQL database. This is called using AGI. The first one has got about 200 rows in it each one is a caller ID number and a name. If I get a match then I set CLI to "name <number>" The second table has over 10,000 rows in it. It has all the telephone exchanges in the UK in it with the prefixes for the exchange. If I

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

Hi All I have a column/variable called time difference. It has a whole list of numbers from 0 through to the hundreds eg 236. I want to assign a corresponding "name" to each variable from a predefined list: Month or less, 1 -2 months, 2-3 months etc So the result would look something like: Time Difference Month 1 Month or

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

Hi, does anyone know of * being used with icecast in any way. Does * support vorbis at all? can anyone who is working on this give me a shout. ---- Shidan

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I will be out of the office starting 11/03/2006 and will not return until 14/03/2006. I will respond to your message when I return. However if you have sent data for processing, please send to operator@cnm.co.uk. Also if you have any urgent queries, please contact Operations on 01924 888700. Thankyou. ________________________________________________________________________ This E-mail

I've been asked to purchase a gsm gateway for use with our asterisk server (for our use, not reselling) I have a spare ISDN port on the server, so I have use either a PRI or VOIP gsm gateway. What would people recommend ? Has anyone used the QuesCom 400 ? I would also love to know a rough idea of cost ;) Once I've gotten the info, I'll post a message on the biz list for a