Re: [asterisk-users] Verification number / code

2009-11-21 Thread Thomas Perron
that is a bit heavy for me. how about some simple way to announce a random number. using RAND. and saydigit exten => s,1,Set(junky=${RAND(1,8)}) On Sat, Nov 21, 2009 at 7:20 PM, Steve Edwards wrote: > On Sat, 21 Nov 2009, Thomas Perron wrote: > > > I want to distribute a random number to eac

Re: [asterisk-users] music on hold

2009-11-21 Thread C F
On Thu, Nov 19, 2009 at 10:31 PM, wrote: > hello friends i want very simple thing in my dial plan. > > 1.When ever calls come at exten 2000 and if it is not answered with in 60 > secs it should hangup. Set absolute timeout to 60 seconds. > > 2.when ever call comes at exten 2000 and if it is an

Re: [asterisk-users] Verification number / code

2009-11-21 Thread Steve Edwards
On Sat, 21 Nov 2009, Thomas Perron wrote: > I want to distribute a random number to each of the first 100 callers to > my IVR. This random number will be matched to their telephone number. > Where in Asterisk can I do this? And, how? > > Logic. > > Call arrives. > Context for announcement begin

[asterisk-users] Verification number / code

2009-11-21 Thread Thomas Perron
I want to distribute a random number to each of the first 100 callers to my IVR. This random number will be matched to their telephone number. Where in Asterisk can I do this? And, how? Logic. Call arrives. Context for announcement begins. You will receive a authentication code at the end of the

Re: [asterisk-users] DIDs

2009-11-21 Thread Thomas Perron
thanks On Sat, Nov 21, 2009 at 12:26 PM, Steve Edwards wrote: > > Thomas Perron wrote: > > > >> I have two DID numbers. I want to configurate my IVR to initiate a > >> context 1 if I dial DID 1. If DID2 is dialed then start with context 2. > > If the DIDs are from different providers, you can s

Re: [asterisk-users] DIDs

2009-11-21 Thread Steve Edwards
> Thomas Perron wrote: > >> I have two DID numbers. I want to configurate my IVR to initiate a >> context 1 if I dial DID 1. If DID2 is dialed then start with context 2. If the DIDs are from different providers, you can specify different contexts in [iax|sip].conf. On Sat, 21 Nov 2009, Alex Ba

Re: [asterisk-users] How to change outgoing DTMF frequencies on zaptel?

2009-11-21 Thread Zeeshan Zakaria
Hi, I am generating the tones from Asterisk, using senddtmf and option D in the Dial command. Is there no way to change it? Can I somehow modify it in tonezone.c and recompile Asterisk? -- Zeeshan A Zakaria On Sat, Nov 21, 2009 at 8:56 AM, Tilghman Lesher wrote: > On Friday 20 November 2009 1

Re: [asterisk-users] DIDs

2009-11-21 Thread Thomas Perron
Hi Alex, Thank you Tom On Sat, Nov 21, 2009 at 10:24 AM, Alex Balashov wrote: > Thomas, > > Thomas Perron wrote: > > > I have two DID numbers. I want to configurate my IVR to initiate a > > context 1 if I dial DID 1. > > If DID2 is dialed then start with context 2. > > Assuming that the DID or

Re: [asterisk-users] Connect two Asterisk Server in IAX ?

2009-11-21 Thread Sylvain MEYNELLY (NEWTEK)
Title: Hi If you want to exchange call between two server I will do the following iax svr1 register => svr2:12...@192.168.0.20 Hi You need to have a user section that authenticate svr1 on svr2 and svr2 on svr1 Phibee Network Operation Center a écrit : Hi My first post get no answer

Re: [asterisk-users] DIDs

2009-11-21 Thread Alex Balashov
Thomas, Thomas Perron wrote: > I have two DID numbers. I want to configurate my IVR to initiate a > context 1 if I dial DID 1. > If DID2 is dialed then start with context 2. Assuming that the DID originator sends you the number in the Request URI, you can just treat them like "extensions" in

[asterisk-users] DIDs

2009-11-21 Thread Thomas Perron
I have two DID numbers. I want to configurate my IVR to initiate a context 1 if I dial DID 1. If DID2 is dialed then start with context 2. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRI

Re: [asterisk-users] How to change outgoing DTMF frequencies on zaptel?

2009-11-21 Thread Tilghman Lesher
On Friday 20 November 2009 17:03:20 Zeeshan Zakaria wrote: > Hi, > > I am having this issue that with one of the Asterisk servers, on zaptel > hardware, that DTMF tones are 10-30 Hz too high than the upper limit for > any DTMF digit frequency. This is causing problem with the equipment on the > oth

[asterisk-users] PCI analog cards on * vs. Quintum

2009-11-21 Thread Sasa Bobek
What is the verdict? There was one positive response, but would like to hear a few more. In addition, what I am looking at is FXO ports to be used with GSM gateways, so any recommendations for specific cards are welcomed. From my experience with PRI cards, I am a little biased toward Sangoma. T

Re: [asterisk-users] can't call through voip provider

2009-11-21 Thread Landy Landy
Hello. I have my server running for about 30 days. Every time I did some changes to my sip.conf file I did reload in the cli. I thought this would change the new values. Somehow it wasn't. I decided to do a restart now and that used my new settings. The same settings I've been posting here the