Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support

Hi Doug, May be you can try a PCI-E card that has a PRI port and asterisk on itself and eliminating the need to install asterisk. www.positrontelecom.com have these cards. Thanks, Krishna On Fri, Jul 8, 2011 at 8:12 PM, Doug Lytle supp...@drdos.info wrote: Patrick Lists wrote: With

Re: [asterisk-users] How to get DTMF in Konference module in Asterisk

Hi Virendra, Set DTMF option in the Makefile to 1 and then recompile/install the app_konference module. Thanks Krishna On Tue, Jun 7, 2011 at 1:31 AM, virendra bhati virbh...@gmail.com wrote: Hi List, I am trying to get DTMF into conference room. for conference I am using Konference

[asterisk-users] DTMF issue in app_konference using with asterisk 1.8.3.2

Hi, I have a requirement where the DTMF entered by a member in konference is passed on to the other members. But the DTMF is not being recognized, when checked the events from manager API, I do see DTMF event being passed, but some how it is not passed to other members. This tells me - may be

Re: [asterisk-users] 4 Port FXO interface

Hi Eric, Wondering if this is something you would like to Try. V114 from Positron Telecom, which supports 4 FXO ports and 1 FXS port. It has asterisk on the card, which would mean you do not need a PC and can install this card as a PCI card on an existing system/server. They also offer an

[asterisk-users] Multicast RTP Paging

HI Guys, I am trying to use the RTPPage application on asterisk 1.4 using the Snom 320's?? My goal is to do the paging using a multicast IP address. I tried the app_rtppage.c and i can only do unicast on the snom's and i was unable to do a multicast.

[asterisk-users] Recording the Calls to a USB Drive

Hi Guys, Merry Christmas and Happy new Year. I am looking for some assistance from the group as i think this might already have been tried before. i have an asterisk server with a external USB Harddisk Drive, just to store recordings. I am using the mixmonitor application for doing the

Re: [asterisk-users] Help with asterisk and avaya SIP trunking

: On Tue, Nov 11, 2008 at 4:56 AM, Krishna Sumanth Chava [EMAIL PROTECTED] wrote: HI Shaun and Robb, Thanks for the assistance. I was able to force the codecs and had avaya talk in the right way. Also addressed the DTMF issues. Glad to hear it. I seem to be having issues with asterisk

Re: [asterisk-users] Help with asterisk and avaya SIP trunking

. I will try connecting the avaya Analog and Avaya IP Phone to IP Office and see if that makes any difference. Thanks again. Regards Krishna On Mon, Nov 10, 2008 at 12:04 AM, Shaun Ewing [EMAIL PROTECTED] wrote: On Mon, Nov 10, 2008 at 2:28 PM, Krishna Sumanth Chava [EMAIL PROTECTED] wrote

Re: [asterisk-users] Help with asterisk and avaya SIP trunking

for detecting Hangups. Please advise. Any help is appreciated as i am new to avaya IP office and am much familiar with asterisk. Regards Krishna On Sat, Nov 8, 2008 at 12:28 PM, Robert Boardman [EMAIL PROTECTED]wrote: Krishna Sumanth Chava wrote: HI Robb, I had the checked the IP Office

Re: [asterisk-users] Help with asterisk and avaya SIP trunking

Krishna On Fri, Nov 7, 2008 at 2:59 PM, Robert Boardman [EMAIL PROTECTED] wrote: Krishna Sumanth Chava wrote: Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya

[asterisk-users] asterisk - avaya ip office SIP trunking

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I

[asterisk-users] Help with asterisk and avaya SIP trunking

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get

[Asterisk-Users] direct a call to a busy channel

Hi, I am looking at a scenario where i have a person on an extension for an indefinite amount of time and i would like to see if there is anyway that i send a call to that extension and to the person by giving a beep sound and then the voice should be placed in that specific channel. The channel

Re: [Asterisk-Users] PRI problems again - What should I do ?

Hi Julian, I think the Dell poweredge2850 servers are not too compatible with the zaptel cards.. Thanks krishna On 11/24/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and down.

Re: [Asterisk-Users] Problems with DTMF on Polycomm Phones

. tone.dtmf.offTime=50 tone.dtmf.chassis.masking=0 tone.dtmf.stim.pac.offHookOnly=0 tone.dtmf.viaRtp=1 tone.dtmf.rfc2833Control=1 tone.dtmf.rfc2833Payload=101/ thanks krishna On 11/7/05, Doug [EMAIL PROTECTED] wrote: At 15:36 11/7/2005, Krishna Sumanth Chava wrote:hi,Would like to have help

[Asterisk-Users] Help with SER

Hi, i am trying to set up SER and Asterisk... I am new to SER and am having problems registering users to SER. i had the Default user Admin registered to my softphone but am unable to register my other users.. i am using mysql with SER.. i saw the difference between the admin user that ihad

[Asterisk-Users] Problems with DTMF on Polycomm Phones

hi, Would like to have help in fixing the DTMF problem i amfacing on Polycomm Soundpoint IPPhones I am having the following network setup.. I have my Asterisk PBX server connected to the Cisco 3620 Router with an ethernet cable whichinturn is connected with a T1 circuit to my SIP Provider.. i