similar to: SET EXTENSION

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

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 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 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 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
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
2006 Mar 14
1
Using up2date to download channels on rhn
Hi all, i have a problem with centos's up2date tool. I have setup a central update server with yam (http://dag.wieers.com/home-made/yam/) under CentOS-4. When yam calls t up2date to download updates, up2date returns this error: Traceback (most recent call last): File "/usr/bin/yam", line 1099, in ? main() File "/usr/bin/yam", line 983, in main
2013 Apr 02
1
R doesn't recognize utils functions, such as arrayIndex( )
Hi all, When I called arrayIndex(20:23, dim=c(4,3,3)), it says "Error: could not find function "arrayIndex"in R". So I called ls("package:utils") to see the functions inside: [1] "?" [2] "adist" [3] "alarm" [4] "apropos" [5] "aregexec" [6] "argsAnywhere" [7] "arrangeWindows" [8]
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 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.
2008 May 08
1
MOH and Licensed G729 codec
Hello All, Recently, I build three Asterisk 1.4 box and installed licensed copy of G729 codec. Before installing the G729 codec I tested the MOH on all three Asterisks box and it was working fine. So I install G729 codec and retested MOH and it was all wavy... Meaning the music was going up and down and missing bits and pieces and choppy... Any idea what did I do wrong? The MOH files are the
2006 Jan 20
1
How to Clear SIP Channels
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
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...
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 "$agi->say_time();" which executes "SAY TIME". Now the current timezone set on the system is "PST", and I have a request to
2007 May 17
2
Blacklist
Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${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.
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
2004 Aug 05
4
newest up2date rpm
i updated to the latest up2date rpm.... then when updating to the latest kernel this is what happened after i ran up2date -fu for the kernel/kernel-source updates Testing package set / solving RPM inter-dependencies... Traceback (most recent call last): File "/usr/sbin/up2date", line 1174, in ? sys.exit(main() or 0) File "/usr/sbin/up2date", line 772, in main