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On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] <all.eforums at gmail.com> wrote: > Hello All, > > Probably a silly question, but we're wondering if people have had any > experience and have data to demonstrate if the performance of the Asterisk > system might suffer in terms of latency etc. if we're to have it retrieve > sound...

Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten => 5150,1,Answer() same => n,Set(CHANNEL(language)=en_AU) same => n,AGI(testagi.sh) same => n,Hangup console output: -- Executing [5150 at AllPhones:1] Answer("SIP/PBX-00000024", "") in new stack

Hello All, Probably a silly question, but we're wondering if people have had any experience and have data to demonstrate if the performance of the Asterisk system might suffer in terms of latency etc. if we're to have it retrieve sound files from a database using odbc as opposed to storing them locally on the filesystem. Note, these are not prompts...these are sound files that are being

.../html; charset=UTF-8 Content-Transfer-Encoding: quoted-printable <div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On M= on, Jan 14, 2013 at 9:36 PM, A E G <span dir=3D"ltr">&lt;<a href=3D"mailto:= all.eforums at gmail.com" target=3D"_blank">all.eforums at gmail.com</a>&gt;</spa= n> wrote:<br> <blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p= x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><fon...

Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER

Hello All, This is probably another one of those completely silly questions that I'm going to hit myself later on, but I have the simplest issue right now but I can't figure out why it's happening: I have a trunk from one * box (box a) to another * box (box b) the call comes in from box a with an extension 2222 which acc. to the peer config in sip.conf is set to use context

On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards <asterisk.org at sedwards.com>wrote: > On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards <asterisk.org at sedwards.com> >> wrote: >> >> I strongly suggest using an existing library for the language of your >> choice. >> > > On Mon, 6 Jun 2011, A E [Gmail] wrote: > > Copy that. Not planning to

Hi All, new to the list. Wondering if anyone has / knows of, a good rate importer tool that can be used to standardize and normalize the ratesheets / rate decks etc. obtained from various carriers so they can be analysed and imported into a DB or be saved as a CSV or something? Thanks so much in advance aeg -------------- next part -------------- An HTML attachment was scrubbed... URL: