Re: [asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

Check it out, thank you. 3 апреля 2012 г. 20:27 пользователь Niccolò Belli написал: > Hi, > If someone is interested I made Debian Squeeze Packages: > http://www.linuxsystems.it/**2012/04/asterisk-1-8-11-0-** > debian-squeeze-packages-with-**t-38-gateway-queue-hints-and-** > fixed-rfc4235/

Re: [asterisk-users] fax tone testing

On 04/02/2012 03:18 AM, Anita Hall wrote: Hi I suspect that my telco set-up is acting funny and I want to use spectral analysis to confirm the culprit :) What is the best way to generate Fax tones from a dialplan and then record them at the other end? Also, where can I get a list of the all the

Re: [asterisk-users] extending fallback numbers

Hi Matt So simple it's amazing!!! Thanx :-) Paolo PS - I forgot to mention it works too... On Tue, Apr 3, 2012 at 9:13 AM, Matt Riddell wrote: > On 3/04/2012, at 6:42 PM, Paolo Supino wrote: >> Hi >> >> Isn't there a better way of doing it that, other than hard coding the >> extension

Re: [asterisk-users] Max number of PCIe cards

Hi Olivier, On 04/03/2012 10:31 AM, Olivier wrote: For training sessions, I'm evaluating the possibility to use a single physical server to host 5 virtual servers, each with its own Dahdi PCIe card, instead of using 5 physical machines, hoping a single physical server would easier to transport,

Re: [asterisk-users] meetme timeout if only one participant

Don't think so. You can set up in the dialplan to skip meetme if the count is 0 or use meetmeadmin to kick out the user when he/she is the last one. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton Sent: Tuesday, April 03

[asterisk-users] meetme timeout if only one participant

Is it possible to have a meetme conference timeout (and go to the next line in the dialplan) if there is only one participant left? Thanks, Matt -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Asterisk 1.8.11.0 Debian Squeeze packages with T.38 gateway, queue hints and fixed RFC4235 (notifycid=yes)

Hi, If someone is interested I made Debian Squeeze Packages: http://www.linuxsystems.it/2012/04/asterisk-1-8-11-0-debian-squeeze-packages-with-t-38-gateway-queue-hints-and-fixed-rfc4235/ Niccolò Il 30/03/2012 17:22, Niccolò Belli ha scritto: http://www.linuxsystems.it/2012/03/new-t-38-gateway-p

Re: [asterisk-users] Voicemail crashs asterisk

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Tuesday, April 03, 2012 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail crashs asteri

Re: [asterisk-users] Voicemail crashs asterisk

On Tuesday 03 April 2012, Steven Howes wrote: > On 3 Apr 2012, at 16:42, Vik Killa wrote: > > #disasterisk fail > > > > #freeswitch win > > #unhelpful comment Yes, but what else do you expect from someone who can't even be bothered to press "page down" a few times so as to put the reply *after*

Re: [asterisk-users] Voicemail crashs asterisk

On Tue, Apr 3, 2012 at 8:42 AM, Vik Killa wrote: > #disasterisk fail > > #freeswitch win > I suppose if you're too stupid to use real Asterisk, that's one alternative. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth

Re: [asterisk-users] Voicemail crashs asterisk

On 3 Apr 2012, at 16:42, Vik Killa wrote: > #disasterisk fail > > #freeswitch win #unhelpful comment S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Voicemail crashs asterisk

#disasterisk fail #freeswitch win >> Hi guys, >> I have the following problem: >> My System: asterisk 1.8.11.0 on debian squeeze >> I login to my mailbox from voicemailmain. >> Once I am logged in, I get my number of messages announced. >> I change the directory of old messages 2 times in a row:

Re: [asterisk-users] process_sdp: Multiple audio streams are not supported

No, there is no way for multiple audio streams to be supported. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org - Original Message - > From: "cjwstudios" > To: "Asterisk Users Ma

Re: [asterisk-users] Voicemail crashs asterisk

Thomas: Please open an issue on the issue tracker, at https://issues.asterisk.org/jira. In the issue, please attach a backtrace generated using the instructions here: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Note that you may need to recompile Asterisk from source with the

[asterisk-users] Voicemail crashs asterisk

Hi guys, I have the following problem: My System: asterisk 1.8.11.0 on debian squeeze I login to my mailbox from voicemailmain. Once I am logged in, I get my number of messages announced. I change the directory of old messages 2 times in a row: 2 1 2 1. Asterisk exits completely. Can your help ple

[asterisk-users] process_sdp: Multiple audio streams are not supported

Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've

[asterisk-users] Max number of PCIe cards

Hi, For training sessions, I'm evaluating the possibility to use a single physical server to host 5 virtual servers, each with its own Dahdi PCIe card, instead of using 5 physical machines, hoping a single physical server would easier to transport, more quiet and cheaper to provision and maintain.

Re: [asterisk-users] concurrent channels limit

no, it's a set of script that I'm supposed to update. However the result will be similar. On 02/04/2012 17:42, Israel Gottlieb wrote: are you by chance using the a2billing script? -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] extending fallback numbers

On 3/04/2012, at 6:42 PM, Paolo Supino wrote: > Hi > > Isn't there a better way of doing it that, other than hard coding the > extension groups? >> >> exten => _20XX,1,Dial(SIP/${EXTEN},30) >> exten => _20XX,n,Dial(SIP/2000,30) Change to: exten => _2XXX,1,Dial(SIP/${EXTEN},30) exten => _2XXX,n