similar to: Blacklist

Displaying 20 results from an estimated 1000 matches similar to: "Blacklist"

2007 Jun 20
1
Asterisk RealTime
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
2007 Aug 11
1
LumenVox Speech Recognition
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
2007 May 23
16
WiFi SIP phones
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
2005 Jul 18
5
TDM04B - Takes long to initialize...
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
2007 Jun 24
3
Nokia N95 + Dial Plan
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
2005 Mar 26
5
Click-to-Talk with Asterisk?
Hi Nitesh, Take a look at this http://www.microappliances.com/site/html/index.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,
2005 Feb 23
4
Vonage <---> Asterisk Working Config!
Hi Nitesh, check out my config that 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:
2007 Jun 14
4
Que on A2Billing
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
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
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...
2007 Jul 19
2
Upgrade Procedure
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
2008 Jan 18
2
SAY TIME + PHPAGI + Timezone
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
2008 Feb 22
2
AGI / Voicemail Que
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
2007 Jan 05
1
integrating with Asterisk and OpenSER for Voicemail
Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)
2007 Oct 03
1
Asterisk Keep Loosing Registration
Hello All, For some odd reasons my Asterisk is keep on loosing registration of my SIP devices. On the SIP device it shows I am RESISTED but when I do "sip show peers" it shows my sip endpoints are "UNREACHABLE". And it keeps on flapping "Peer '9099993456' is now UNREACHABLE!" and "Peer '9099993456' is now REACHABLE!"... I changed my
2010 Mar 14
2
dahdi-linux-complete-2.2.1+2.2.1 failed to compile
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
2006 Oct 12
2
Some file aren't loaded its No file in that Directory.
Hello Users, I Installed the Asterisk-1.2.11, For My Real time Use I'm Use MySql For Asterisk Database, By Using the Asterisk-addons -1.2.4 in My Linux. For My Voice messages Storage , I want To Use the MySql. In Googled it shows me the ODBC integration.. Is it need for that ODBC integration with MySql for my Voice Message storing in MySql. When I'm trying to integrate with ODBC +
2007 May 23
3
What replaces SetCallerPres in 1.4
Hello SetCallerPres function seems to be removed from Asterisk 1.4. What function or application replaced it? Bit of a problem if you want to use CLIR on your PRI connections. Jon No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.467 / Virus Database: 269.7.6/815 - Release Date: 22-05-2007 15:49 -------------- next part -------------- An HTML
2005 Feb 18
2
VONAGE <----> ASTERISK SIP TERMINATION?????
Has anyone out there successfully set up their * box to terminate their VONAGE calls? I (and I am sure lots of others) would love to hear how you did it. I'd like to be able to get rid of the extra hardware I have hanging around here and use the ASTERISK machine to handle the SIP termination instead of needing to have a Linksys modem (w/phone) and an additional X100P card. Thanks.
2006 Nov 15
1
How to do the Call Snooping
Hello Users, I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping, I seen that " What is Trixbox " in Asterisk I Use only some Feature in Asterisk (20), Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk Server Help me please :P -- Thanks and Regards Ravi Prakash Sunkara
2004 May 18
1
VoiceMailMain dumps user back into my incoming context after leaving a message
I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang