search for: goksi

...h 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

...use g729 in passthru mode. and I guess disabling hyperthreading will save me more system resouces. I will be glad, if you can give me more info on system management cos i think with that system, it should able to handle at least five E1's. I say thank you for finding time to reply my mail. goksie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060530/b03ecc2a/attachment.htm

...ll 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. Goksie

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 >> >> -- >> _____________________________________________________________________ >> -- B...

Hi I have created a rudimentary perl script that does most of what I want but occasionally in seems that a file will not play. I see the message getting sent to Asterisk but no reply to say that it completed. In fact, the very next SAY DATE command works and everything after that but the previous message seems to be ignored. In addition, I found in one case that I had to read to STDIN

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

...=> 65 enabled => no [link-l4] linkset => mahmud channels => 1-31 schannel => firstcic => 97 enabled => no [host-B] enabled => yes opc => 98 dpc => mahmud:998 links => l1:1 Can someone advice on the way forward or has anyone implemented chan_ss7-1.0.0. Thanks Goksie

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

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.

...oh323.conf [abcd] type=user context=default ip=62.193.1XX.2XX disallow=all allow=gsm allow=ulaw this gateway can make call, even if these lines are commented out and you restart the asterisk bcos of that. yet the ooh323 doesn't do any authentication like sip channel does. can someone help. goksie

hi, I am installing the asterisk-1.2.19 with zaptel-1.2.8 on FC5. I got install installed ok.. after i had disable the xpp_usb module. However, when i try to compile asterisk and having this error I will be glad for your kind response. Goksie "chan_zap.c: In function ?pri_dchannel?: chan_zap.c:9203: error: ?pri_event_setup_ack? has no member named ?call? make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.19/channels' make: *** [subdirs] Error 1"

Can someone guide me on what package I can use to do least cost routing in asterisk-1.4 without going through the prepaid calling card platforms. I have tried Asterisk::LCR and LCDial without success, if more help on either too. I will be glad. I will be glad for good pointers. Thanks. Goksie

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>

...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. i have asterisk-1.12.1 zaptel-1.9.1 chan_ss7-0.8.4 Goksie

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

Hi, To solve echo problems, I'm considering 2 alternatives. 1> Sangoma A104d - I can't find support for asterisk 1.2.1 2> Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command "reload" can change rx_gain and

...19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event: Alarm cleared on channel 9 Mar 1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event: Alarm cleared on channel 10 Mar 1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event: Alarm cleared on channel 11 For all the 31 channels goksie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060301/4cdf79d6/attachment.htm

..._ = "__ast_pbx_run" #6 0x0809386c in pbx_thread (data=0x0) at pbx.c:2505 No locals. #7 0x00c161d5 in start_thread () from /lib/tls/libpthread.so.0 No symbol table info available. #8 0x00a972da in clone () from /lib/tls/libc.so.6 No symbol table info available. (gdb) (gdb) (gdb) quit goksie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060503/952f22fe/attachment.htm

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>

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