similar to: asterisk admin interface

On May 16, 2016 22:15, "Telium Technical Support" <support at telium.ca> wrote: > > You don't mention a configuration generator (like Elastix/FreePBX) so I'll > assume you are using a plain old vanilla Asterisk installation. In which > case all user/endpoint information is kept in config (ini) files, and no > user/endpoint manipulation is done through the

Please don't top post. > On Mon, 16 May 2016, Goke Aruna wrote: > can anyone give me a guide on any asterisk admin solution / interface > for config management, and monitoring? No database use is intended and I > prefer open source. > On May 16, 2016 20:50, "Steve Edwards" <asterisk.org at sedwards.com> > wrote: > > Can you be a bit more specific

Wow! I've been *-ing for about 6 years and had literally no idea about that! I can see a way I could put it to a different use, but it seems to be a bit of a sledgehammer to crack the walnut of "how many current callers" compared to one line of (albeit hacky) dialplan. That's making me sound ungrateful. I don't mean to be! On Sun, 14 Jun 2020, 22:39 Steve Edwards,

Hello all, Can someone recommend what hardware to use for a 1000 analogue line capacity asterisk PABX? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/2bcd322f/attachment.html>

hello all, I am looking for an implementation of a 10 man call center. low cost license or GPL will be preferred. I will be glad for your help. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160622/8e2e5ab8/attachment.html>

Thank you... but "just update the database" - hmm, what database? Did you mean ARI? I still can't find the command! The asterisk wiki is somewhat, um... spread around! On Sat, 13 Jun 2020 at 16:56, Steve Edwards <asterisk.org at sedwards.com> wrote: > > On Sat, 13 Jun 2020, Jonathan H wrote: > > > I need to ensure that a MusicOnHold stream is only running when

Thanks Harry. I will check and revert. I hope it work perfectly with asterisk. Regards On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor <hmcgregor at biggeeks.org> wrote: > Hi, > > For analog, I really like telco grade channel banks. > > I would recommend the adit 600, there is a good market on Ebay, and you > can do 48 channels per adit 600, with 2 T1 interfaces. Having

hi all, Can someone help me with a kind of howto build call center around asterisk with all the necessary features like CTI, call recordings, call spying, real time monitoring etc? I will be glad if it is an open source code. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.

Thanks Mitul, The server spec is okay but I need information on the fxs hardware to use. Regards On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani <mitul at enterux.in> wrote: > Quad core Xeon with 4GB ram > On Feb 17, 2016 12:32 PM, "Goke Aruna" <goksie at gmail.com> wrote: > >> Hello all, >> Can someone recommend what hardware to use for a 1000 analogue

Hello all, I have a set up that answer my customer. and its working well, however, the number of call to technical dept is what i want to reduce. I want all call to get to voice prompt except that that enter when minutes is 15, 30, 45, 60(in multiples of 15 minutes). how can i achieve this and what application can i use to get this done. I will be glad, if someone can give me a hint on this.

Thanks. The configure run successfully. but I got the warning below.. checking for the ability of -lsrtp to be linked in a shared object... no configure: WARNING: *** configure: WARNING: *** libsrtp could not be linked as a shared object. configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr configure:

On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote: > Sangoma 50 port FXS Thanks. Will I now stack 20 boxes in order to achieve the 1000 FXS lines? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217/0d4c2800/attachment.html>

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

hi all, can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2 on it with options as stated below - ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/ --with-pjproject-bundled when I tried to run "make menuselect". i get the error below. Makefile:109: makeopts: No such file or directory **** **** The configure script must be executed

On Fri, 18 Mar 2016, Trey Hilyard wrote: > I thought this would be as easy as > exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10}) Have you tried the '_!.' pattern? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.

I'm parsing ` sudo asterisk -rx "core show calls" | grep active | head -c 1 ` as an external call from within the Asterisk dialplan then passing it to agi, but this seems really hacky and ugly. However, I cannot find any ARI/AGI/AMI function (or global variable I can get with agi) which shows me this. Any ideas?!? In case it helps and you're wondering why... I need to ensure

I'm attempting to setup Asterisk 1.4.0 CDRs to use MySQL. Modules show like cdr_mysql.so tells me it is loaded. Reload cdr with MySQL started or stopped makes no difference in the errors. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070220/55700f37/attachment.htm

with the slip of a finger, i destroyed by extensions.conf (grep -i > extensions.conf) I have a backup that is dozens of hours of code old. is there a way i can use the asterisk cli (or some other asterisky method) to recreate that extensions.conf ?