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hi all, 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. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

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>

...e per 2 channel banks. > > You could service this with six eight port T1 cards, or with eleven/twelve > quad T1 cards. I would distribute across two, three, or even four servers > for redundancy/resiliency and load balancing. > > -Harry > > > On 02/17/2016 12:16 AM, Goke Aruna wrote: > > > 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 > > > > > -- > ________________...

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 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>

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 line >> capacity asterisk PABX? >> >> Regards >> >> -- >> ___________________________________________________________________...

...RNING: *** If you do not need SRTP support re-run configure configure: WARNING: *** with the --without-srtp option. i hope i am not too far away. Regards On Sun, Feb 12, 2017 at 5:46 PM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Sun, Feb 12, 2017 at 05:24:16PM +0100, Goke Aruna wrote: > > 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 > &g...

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.

...ial NaN, *not* for NA */ are declared leaving out R_IsNaNorNA. This funcion was declared in R-1.9.1. So I changed (R_IsNaNorNA) to (R_IsNA) in ~/Rggobi/src. Then I built Rggobi and installed it using R CMD Can anyone tell me if there is any other way to do it? Was I right? Thanks, Sincerely, Aruna Buddana, Intern, Statistics and Data Sciences Oak Ridge National Laboratories.

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

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.

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: >...

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>

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

...embarrassed to have a setup like that tied to my name if I was a consultant etc. Someone will come in after you and ask who set it up and the customer will say you :) On Feb 17, 2016 4:23 AM, "A J Stiles" <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Goke Aruna wrote: > > Hello all, > > Can someone recommend what hardware to use for a 1000 analogue line > > capacity asterisk PABX? > > > > Regards > > A PCI express card with four primary rate ISDN ports, each linked up to a > channel bank, will give you 120 analogue l...

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

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

can someone overthere help? the server specs are as follows HP DL380G4 Dual Intel Xeon 3.2GHz processor with 4GB RAM, running fedora core 3 asterisk-1.2.5 ss7-0.8.3d. using sip as advised to receive calls from another gateway in US. using g729 in transcoding way. however, I noticed the call hit the 51 active calls which is 102channels, I run "top" to check the system resources usage

Hi, I have asterisk 1.0.7 and spandsp-0.0.2_pre18. txfax return a non-zero return code only if the fax file is not found. Unfortunately I can't get any information, whether the fax was transmitted completely or not. Will an update to a newer version change this? Thanks for telling me your experience! Roger.

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I