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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

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

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,

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

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, 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

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

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.

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:

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

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

Hi, all I am in Australia and I have to setup Asterisk in few offices. There will be IP phones in each office and I must be able to call between offices. I need actual handsets. I need "standard" handsets to be used by people. Those must support features like CID, call forward, etc. --- your normal office feature set. Also I need some sort of more complex handset to be used by

Hello All, I am trying to build an IVR based survey system and I would like to save the key touch-tones to MySQL DB. It is a simple application where it determines the Caller ID and asks couple of questions and save the Callers input to the MySQL DB. I am thinking of building IVR using FreePBX, but my main worry is how to save the Callers input? Simple flow: - 1) Verify the CallerID, if not

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, I am almost done with my notifications system, but I am stuck with prompting the correct time. I went over the "phpagi" doc's, on how to say a given time using "SAY TIME <time> <escape digits>. According to http://www.voip-info.org/wiki/view/say+time it say <time> is number of seconds elapsed since 00:00:00 on January 1, 1970, Coordinated

Hello All, I don't understand where is the problem... I have Callback setup and it works fine when tested within US. Works fine meaning the DTMF tones are passed when prompted to enter the phone number. But when I test with some international countries, callback works but DTMF tones are not passed... Is it: - a) Asterisk problem? b) Callback problem? c) VoIP provider problem?

Hello All, I have my AGI scripts streaming wav files and I would like to increase the volume on it. Is there any way to increase the volume on outbound SIP trunks, instead of me changing the wav file one by one? Please help? 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