search for: divecha

Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh

Greetings list, What are people's experiences with WiFi SIP phones? When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices. I assume things must have moved on somewhat since

Hello All, I got my TDM04B card installed and configured. Everything works fine I can receive calls and route to appropriate extensions. The only problem I am facing is Slowness. When I dial the PSTN number which is connected to Zap 1-1 after two ring it answers and then run the AGI script. What I did was assign it to a specific extension. So all inbound call on that PSTN number should

...dex.php?section=Products&page =clienthowto.php I've never implemented it though so I would appreciate some feedback on if it works. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Saturday, March 26, 2005 3:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Click-to-Talk with Asterisk? Hello All, Is there any open source Click-To-Talk feature which we can integrate with Asterisk PBX? It's a button created on a w...

Hello All, Recently I added some Nokia N95 customers and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

...I have for the Faktortel config in the asterisk@home sourceforge forum, you'll probably be able to work out how to set it up from there. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, February 23, 2005 4:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Vonage <---> Asterisk Working Config! Hello All, I know this question has already been answered but for some reason I am failing to receive inbound call...

Hello All, I have my own AGI script running and I am trying to push the call to voice mail when Busy, Unavailable and Not Answered. Everything is working fine but the only problem is voice mail greetings for Busy and Unavailable is not played. By default only "Temp Greetings" voice mail greetings is played. I am passing the correct parameters for Busy => 'b', Unavailable

Hello All, Is there any way to change the timezone on the fly? I have this little time clock program running on Asterisk system developed using PHPAGI. Currently, whenever user logs in, Asterisk will prompt the current system time using "$agi->say_time();" which executes "SAY TIME". Now the current timezone set on the system is "PST", and I have a request to

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Hello All, I would like to upgrade my recently installed Asterisk 1.2.21.1 to Asterisk 1.4.8? My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6 05:25:07 EDT 2007 i686 i686 i386 GNU/Linux Is there any detail step by step procedure to uninstall the current version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons 1.4.2? Cheers, Nitesh

Hello All, I'm trying to do a fresh installation on Ubuntu Server 9.10 (Karmic) 64-bit but I am getting error when "make config" is trying to install the init script... Here is the output: - Can anyone help me please... Thanking in advance... Cheers, Nitesh ################################################### ### ### DAHDI tools installed successfully. ### If you have not done

...terisk - safe? (David Gomillion) > 8. Blank Voicemails (Leah Newmark) > 9. Re: Pass Dialed number to a script (Jared Smith) > 10. Re: Parsing IAXPeers from Asterisk Manager (PHP API) (Jared Smith) > 11. Re: G729 copy protection (Jared Smith) > 12. Re: Upgrade Procedure (Nitesh Divecha) > 13. Re: Gtalk/Jabber connect issues in 1.4.8 (Bruce Ferrell) > 14. Re: Upgrade Procedure (Jared Smith) > 15. Re: 1.4.X howto disable able xpp with ./configure (Tzafrir Cohen) > 16. Re: G729 copy protection (Bruce McAlister) > 17. Re: G729 copy protection (Bruce McAlister) &gt...

...TA-based account. Because the softphone account works with openly available soft clients, it also works with asterisk. The big "secret" is that they use port 5061, rather than port 5060. > > I thought Vonage did not allow this? > > > -Randy > > > Nitesh Divecha wrote: > > >Hello Asterisk Users, > > > >After Brain storming for couple of hours, days, and weeks, > finally got > >Asterisk to work with Vonage for Inbound and Outbound calls. > > > >Requirement: - > >1) Vonage Softphone account > >2) Aster...

Yes. http://lists.digium.com/pipermail/asterisk-users/2005-February/087538.ht ml -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh Divecha Sent: Wednesday, March 02, 2005 2:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Call Forwarding to Cell Phone, Pager, etc Hello all, Was just wondering if Asterisk can do Call forwarding to cell phones, pagers, home phone, etc. For example,...

Hello All, Is there any way to clear the SIP Channels? When I run "sip show channels" on CLI I see +500 SIP Channels active with "unknown" codec. But thats false information, because when I restart my Asterisk and run "sip show channels" I will see the actual active channels with correct codec info. Anyways to clear the sip channels without restarting the

Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies". Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "*there's no support for NAT keep-alives

Hello All, Is there any way to pass additional parameters while calling AGI from *.call file? Channel: Local/1000 at from-internal MaxRetries: 0 RetryTime: 15 WaitTime: 15 Application: AGI Data: recordvoice.php Something like Data: recordvoice.php?id=3453&name=asterisk Cheers, Nitesh

Hello All, How can I SET EXTENSION from context? This is my context: - [docall-usa] exten => _NXXNXXXXXX,1,Answer exten => _NXXNXXXXXX,n,Set() ; <<What do I need to set here>> exten => _NXXNXXXXXX,n,DeadAGI(dousacall.php|1) exten => _NXXNXXXXXX,n,Hangup I need to add 1 in front of ${EXTEN} and then send the call to dousa.php. Set(CALLERID(number)=1${EXTEN}) will set