[asterisk-users] Top posting - there is no rule.

What's with the occasional Un-Top-posting, there is no rule that says you can't, http://www.asterisk.org/community/rules My preference is top posting, as you see the answer at a quick glance, instead of reaching for the scroll bar (or whatever key stokes are required) to get to the bottom, to

Re: [asterisk-users] Top posting - there is no rule.

On Sun, Apr 3, 2011 at 1:54 AM, Alec Davis siva...@paradise.net.nz wrote: What's with the occasional Un-Top-posting, there is no rule that says you can't, http://www.asterisk.org/community/rules Really? Why bring this up again? The last 60-some odd thread in January wasn't long enough for

Re: [asterisk-users] Top posting - there is no rule.

Really? Why bring this up again? The last 60-some odd thread in January wasn't long enough for you? Or perhaps you're just out looking to troll?From the page you linked: 5.Responses should be placed under the original quoted text. Sorry my mistake, didn't read to bottom of the

Re: [asterisk-users] BRI detection

On Fri, Apr 01, 2011 at 06:06:45PM +0530, mahesh katta wrote: Hi, I need to configure BRI 4span card in dubai in vicidialnow for dialer perpose. in that i have small confusion which is NT an TE mode . that was i am setting perfectly but dubai telco what they are use for this i dont know

[asterisk-users] [DIGIUM FAX] HANGUP problem

Hi I noticed that all my fax negociated in *V17 *hangup Here my conf fax show version FAX For Asterisk Components: Applications: 1.8.4-rc2 Digium FAX Driver: 1.8.4_1.3.0 (optimized for i686_32) Here an example of failed fax: -- Channel 'SIP/VOIP-OUT-0043' FAX session

Re: [asterisk-users] The SIP channel driver - I'm giving up.

On Fri, Apr 01, 2011 at 09:06:53AM +, Tony Mountifield wrote: In article b4ed7e5e-56d7-4ab3-8f9c-0d120a741...@edvina.net, Olle E. Johansson o...@edvina.net wrote: Friends, After having spent many years working with the Asterisk SIP channel driver and the SIPv2 protocol, I have

[asterisk-users] hello

I have bought a pair of Apple phones on---www.ofenno.com---they have pretty good quality. Here I would like to recommend it to you. Their company is holding a promotion activity now, so you can buy anything you want on it with free delivery charges. There must be anything you like, I hope you

[asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call

Hi i use this into my extension : exten = _00339,1,Set(foo=${SIP_HEADER(To)}) exten = _00339,2,Set(cut1=${CUT(foo,:,2)}) exten = _00339,3,Set(CLI=${CUT(cut1,,1)}) exten = _00339,4,Set(toexten=${CUT(CLI,@,1)}) exten =

Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call

In that situation, I've had to do a pickup macro that kind of primes the audio. Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio)) context start-audio { s = { Playback(silence/1); } } The above might help... What it does is plays an audio track on the callee's channel

Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call

Hi Mark Thanks for your answer, but i am new in asterisk ;=) the context start-audio ... i put it into the extension.conf ? because i have a error: [Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No '=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf [Apr 3

[asterisk-users] From 1.4 to 1.8: stdexten issue

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like

Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call

I gave you the syntax in ael format, if you want to use extensions.conf you'll have to use the syntax that's applicable, which is: [start-audio] exten = s,1,Playback(silence/1) On 04/03/11 14:14, Olivier CALVANO wrote: Hi Mark Thanks for your answer, but i am new in asterisk ;=) the context

Re: [asterisk-users] From 1.4 to 1.8: stdexten issue

Mathieu Chouquet-Stringer math...@csetco.com writes: I've googled and pretty much tried all forms of the syntax but I've yet to make it work. For instance I tried not including stdexten and calling Gosub with (stdexten,${EXTEN},1(SIP/phone1)) but it didn't work either... stdexten in the

Re: [asterisk-users] hello

-BEGIN PGP SIGNED MESSAGE- Hash: RIPEMD160 WTF does that have to do with Asterisk? On 04/03/2011 05:56 AM, ALAEDDINE abbech wrote: I have bought a pair of Apple phones on---www.ofenno.com---they have pretty good quality. Here I would like to recommend it to you. Their company is

Re: [asterisk-users] hello

Bradley D. Thornton wrote: WTF does that have to do with Asterisk? It's called spam. And either he doesn't know what non-commercial discussion means or he signed up just to send this. Doubtful we'll see him again. Doug -- Ben Franklin quote: Those who would give up Essential