[Asterisk-Users] Detection of Answering Machine

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over

[Asterisk-Users] g729 liscence question

I have a TDM400P with one FXS and one FXO.. how many liscence(2) I will have to buy? Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] g729 liscence question

. Thanks again. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 01 Aug 2005 18:19:03 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] g729 liscence question On Mon, 2005-08-01 at 10:53 -0800, Innocent Evil wrote: I have a TDM400P with one FXS

[Asterisk-Users] Asterisk as PSTN gateway, voice mail server with SIP

Hello, I am sure this has been answered so many times as it is one of the most fundamental features of Asterisk. Here is my scenario, I have setup my asterisk server with a TDM400p which have one FXO and FXS card. My SIP server is up and its working fine only in SIP network ( I used ser) For my

RE: [Asterisk-Users] how may channels

keep approx. 32kb per channel.. -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how may channels how many channels using codec g729 can be done by an internet bandwidth to 512kb

[Asterisk-Users] number 'register = ' in sip.conf

how many 'register =' I can have in sip.conf___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] call outside from FXS through FXO

Hi, I am trying to make an outbound call from phone attached to FXS port. My telephone (VoIP) line is connected to FXO port (Zap/4) Default context for channel # 4 is 'directdial' here is part of my extension.conf [directdial] ignorepat = 9 exten = 9,1,Dial,Zap/4/ exten = 9,2,Congestion

[Asterisk-Users] Switch between FXS ports

Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep

Re: [Asterisk-Users] Switch between FXS ports

: [Asterisk-Users] Switch between FXS ports Innocent Evil wrote: Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through

RE: [Asterisk-Users] Switch between FXS ports

PROTECTED] On Behalf Of -Innocent Evil -Sent: Monday, August 15, 2005 2:17 PM -To: asterisk-users@lists.digium.com -Subject: [Asterisk-Users] Switch between FXS ports - -Hello, - -I have two FXS port on my TDM card. -channel 4 is attached with a telco line that I use -frequently

[Asterisk-Users] problem with sound device

I am getting this whenever I start asterisk. Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device: Resource temporarily unavailable This is my sound card: Multimedia audio controller: Fortemedia, Inc Xwave QS3000A I am not sure... what I am doing wrong. Please help.

RE: [Asterisk-Users] Switch between FXS ports

I am clear with this issue. Thanks everybody for answering me. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 15 Aug 2005 10:16:34 -0800 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Switch between FXS ports Hello, I have two FXS port on my TDM card.

[Asterisk-Users] Asterisk and LCR

Hello, How do you guys implement LCR in Asterisk? Thanks,___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Asterisk and LCR

How about this: 1. Put all the routes of all the providers in a MySQL table 2. Write a script with a 'clever' algorithm to find out cheapest route of each prefix. 3. Based on #2.. make a lcr_cheapest_route.conf 4. include lcr_cheapest_route.conf in extension.conf But I don't know, how much

RE: [Asterisk-Users] Asterisk and LCR

Are you saying realtime mysql is not clever? That is exactly what it is supposed to do. BTW, how do you integrate mysql with asterisk? any link, documention, tutorials would be greatly helpful. Thanks,___ Asterisk-Users mailing list

[Asterisk-Users] Asterisk and Port

[EMAIL PROTECTED] ~]# netstat -naptu | grep asterisk tcp0 0 0.0.0.0:20000.0.0.0:* LISTEN 9231/asterisk udp0 0 0.0.0.0:27270.0.0.0:* 9231/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 9231/asterisk udp0

Re: [Asterisk-Users] problem with sound device

On Monday 15 August 2005 21:08, Innocent Evil wrote: I am getting this whenever I start asterisk. Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device: Resource temporarily unavailable sounds like your soundbard is blocked by another program. Sometimes applications like

[Asterisk-Users] Festival error

Hello, I have installed festival (the rpm package came with fc4). But getting this: client(1) : rejected from myserver not in access list whenever I try to access it from asterisk . I found in documentation: If you see a message such as: client(1) : rejected from myserver.mydomain.com not in

RE: [Asterisk-Users] Festival error

Sorry, the file I am talking about is in right place !! but not sure what to add in festival_server script. Thanks, Hello, I have installed festival (the rpm package came with fc4). But getting this: client(1) : rejected from myserver not in access list whenever I try to access it from

Re: [Asterisk-Users] Festival error

going to make my festival server accept connect from anybody? If it is, I dont want to do that. I just want to add my asterisk sever to festival's client list. Thanks, Innocent Evil wrote: Hello, I have installed festival (the rpm package came with fc4). But getting this: client(1

[Asterisk-Users] Festival sounds too wired !!

Hello, I was just able to connect to my festival server.. but the voice generated by festival sounds too wired ..really. I installed only festival, i didn't install speech_tools and couple progams as was documented in voip-info.org How can I tune up festival to have better voice (not as good as

RE: [Asterisk-Users] Searching For a Voip Provider

Please change the subject to 'Advertisement of a VoIP Provider' -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 18 Aug 2005 11:55:50 -0700 (PDT) To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Searching For a Voip Provider Hi: Please advice me of a voip

RE: [Asterisk-Users] VoipJet Problems - anyone?

I noticed their mysql server is down or can't connect to mysql server. I tried to download there cvs format price list. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 18 Aug 2005 16:04:30 -0400 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoipJet Problems -

[Asterisk-Users] Out of G.729 Decoder Licenses!

Hello, I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium website) SIP user (100) is calling another SIP user (101). As 101 is not online, my SIP server is redirecting that call to Asterisk. Asterisk forward it to 101's voice mail box. SIP user 100's phone have g729

Re: [Asterisk-Users] Out of G.729 Decoder Licenses!

, -Original Message- From: [EMAIL PROTECTED] Sent: Fri, 19 Aug 2005 09:02:21 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Out of G.729 Decoder Licenses! Innocent Evil wrote: Hello, I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium

[Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain

Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem is, voice mail box can't read password sent from SIP phone, but I don't have any problem to read password from the handset attached to my asterisk box. Your help will be greatly appreciated.

Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain

of these formats.) Preferably, use RFC2833 or SIP INFO. Find a setting that is available on your phone and on *, and make sure they're set to match. Once you do that, it should work. Jeremy Innocent Evil wrote: Hi, I am using Asterisk cmd VoiceMailMain to manage voice mail. Problem

Re: [Asterisk-Users] Sending digits from SIP to Asterisk's VoiceMailMain

:46 PM, Innocent Evil wrote: my sip phone have dtmf relay: rfc2833 asterisk sip.conf have dtmf relay: rfc2833 in associated context. I tried with Inband.. but g729 doesn't support it. I have g729 liscence from digium I havn't try with INFO yet. I prefer to have rfc2833 as dtmf

Re: [Asterisk-Users] DTMF not working

I am having same problem .. DTMF is not working from a SIP phone while sending to Asterisk cmd VoiceMailMain. Would you please explain this line !941+1336/100,!0/100, /* 0 */ what value is what and how it affect on DTMF tone generation. Thanks, I had a similar problem that seems to be

[Asterisk-Users] AGI + Ruby

I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

Re: [Asterisk-Users] AGI + Ruby

and possibly cheaper too. I believe the current IDE is 12.4K Innocent Evil wrote: I would like to write AGI script in Ruby Would anybody please show me right direction.. Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] DTMF not working

. The 100 is the duration of those tones. The tones are in Hz. I'm not sure what units the duration is in, but I bumped mine from 100 to 400 and that seems to do the trick. The part of the string that reads !0/100 just shuts the tone generator off. Rob Innocent Evil wrote: I am having

RE: [Asterisk-Users] AGI + Ruby

] [mailto:[EMAIL PROTECTED] On Behalf Of Innocent Evil Sent: Wednesday, August 24, 2005 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI + Ruby What IDE are you talking about? Any URL would be helpful. Thanks, -Original

Re: [Asterisk-Users] AGI + Ruby

Subject: Re: [Asterisk-Users] AGI + Ruby Y'see it? There it goes! Right over his head. Huddleston, Robert wrote: U joke - duh! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Innocent Evil Sent: Wednesday, August 24, 2005 3

[Asterisk-Users] where can I get low cost g723.1 liscence

Hello, Would you please suggest me, where can I buy g723.1 liscence in cheap. I might need a liscence for 10-50 channels. Thanks,___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

RE: [Asterisk-Users] AGI and incoming call

Why don't you do something like this: exten = 12345678,1,Dial(10) exten = 45874521,1,Dial(11) exten = 32544884,1,Dial(12) replace Dial(10) and so on with apppriate extension. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From:

[Asterisk-Users] SDT Message Signal

Hello, My phone's VMWI (Visual Message Waiting Indicator) is able to detect SDT message signal. But how I would configure asterisk to send SDT message signal to a certain extension? Thanks, ___ --Bandwidth and Colocation sponsored by Easynews.com --

[Asterisk-Users] International Dialing Code

I am trying to download a list of international dialing codes. Would anybody please post a link to get it Thanks in advance. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] AGI and AUTOHANGUP

Hi, Using AUTOHANGUP, I can force a call duration within a time limit. I would like to playback a message before 1 minute of autohangup. How can I accomplish it? Would anybody please give me right direction. Thanks, You don't have any choice, you already made it before you came

Re: [Asterisk-Users] International Dialing Code

: Innocent Evil wrote: I am trying to download a list of international dialing codes. Would anybody please post a link to get it Google IS your friend. Did you try? http://www.0xdecafbad.com has one in the first few links of the main page/articles page. In asterisk dialplan format as well as csv

Re: [Asterisk-Users] International Dialing Code

you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Sun, 20 Nov 2005 09:27:09 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] International Dialing Code On Sun, 2005-11-20 at 15:20 -0200, Hermann Wecke wrote: Innocent Evil wrote: I am trying

RE: [Asterisk-Users] AGI and AUTOHANGUP

Comeo'n AGI guys.. Please say something. Hi, Using AUTOHANGUP, I can force a call duration within a time limit. I would like to playback a message before 1 minute of autohangup. How can I accomplish it? Would anybody please give me right direction. Thanks, You don't have any

RE: [Asterisk-Users] International Dialing Code

gotten a few more changes as well.. Thanks :) On Mon, 2005-11-21 at 22:03 -0800, Innocent Evil wrote: Lots of country have wrong prefix. Andorra,376 should be 1376 Angola,244should be 1244 Antarctica,6721 should be 1672 http://en.wikipedia.org/wiki

Re: [Asterisk-Users] AGI and AUTOHANGUP

look at the code for ASTCC in the asterisk cvs or look at astpp-callingcard.agi in the cvs code available @ www.aleph-com.net/astpp Darren Wiebe [EMAIL PROTECTED] Innocent Evil wrote: Comeo'n AGI guys.. Please say something. Hi, Using AUTOHANGUP, I can force a call duration

Re: [Asterisk-Users] Truncated CDR records

you can use 'w' option with 'Dial' on 1.2.x I don't think w do anything like 'wait', If I am wrong, correct me someone please According to app_dial.c w- Allow the called party to enable recording of the call by sending\n the DTMF sequence defined for one-touch recording in

[Asterisk-Users] cdr enhancement with 'rate' column

Hello, I am trying to enhance my cdr records. What I am trying to are: 1. Add an option in Dial, Say R. to pass rate (price per minute) for the call to do that I will have to modify app_dial.c 2. Dial would use option R to set cdr 3. I will also need to add one more function in cdr.c, say

Re: [Asterisk-Users] cdr enhancement with 'rate' column

1. Add an option in Dial, Say R. to pass rate (price per minute) for the call to do that I will have to modify app_dial.c 2. Dial would use option R to set cdr 3. I will also need to add one more function in cdr.c, say something like ast_cdr_setrate(...) None of this is necessary.

[Asterisk-Users] AGI + CDR

Hi, I have an AGI script that is called after receving a call on a channel. And my script executel AGI cmd Dial to make another call. Is there any reason not to have CDR record for the call that was initiated in the AGI script? Or I am just missing something basics . Thanks, -- You don't have

[Asterisk-Users] cdr_manager.conf

Hello, While I was trying to get right CDR record from AGI script, I came across cdr_manager.conf I am trying to learn about cdr_manager.conf What is the purpose of cdr_manager.conf? How I can configure it? I did google, really didn't have very good luck. Would anybody please write couple of

Re: [Asterisk-Users] DTMF not working

-Users] DTMF not working Innocent Evil wrote: I am having same problem .. DTMF is not working from a SIP phone while sending to Asterisk cmd VoiceMailMain. Have you set DTMF to out of band RFC2833? In band won't work. At least in my version of HEAD John Novack Would you please

[Asterisk-Users] AGI problem with library path

Hi List, My AGI seems work well in asterisk -vvvc mode, other than that it doesn't work. Its seems to me, when I run asterisk as daemon (service asterisk start .. on fc4), it doesn't know about my library path. How can pass libray path to my AGI script or asterisk?

[Asterisk-Users] problem with FXS module

All of sudden my FXS module is not working. I have a TDM card with one FXS and one FXO, FXO module seems working fine. I also noticed the LED is not on for my FXS module while it is on for my FXO module. Sep 13 12:11:44 WARNING[9870]: chan_zap.c:887 zt_open: Unable to specify channel 1: No such

Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards

My TDM400 on fc4 was working great.. all of sudden ..i am having the same issue ..you guys are having all i tried to run asterisk as non-root user.. and I was able to run it as non-root and was able to receive and send call using asterisk.. I am not sure.. what thing I did wrong and coz all the

Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards

to SuSE or other Linux distribution such as Debian. /Y.T. --- Innocent Evil [EMAIL PROTECTED] wrote: My TDM400 on fc4 was working great.. all of sudden ..i am having the same issue ..you guys are having all i tried to run asterisk as non-root user.. and I was able to run

Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards

como'n folks.. ... Well, as I told earlier.. my asterisk was running great with one fxo and one fxs module of a TDM400P All i tried last night to run asterisk with non-root I must did something wrong while I was trying to do that FXO module on channel # 1 FXS module on channel # 4

Re: [Asterisk-Users] Fedora Core 4 not recognizing X100P cards

just an update for all curious folks I just replaced my FXS card and everything is working great .. I am running asterisk as non-root user. Moreover, I couldn't figure it out how/why my FXS card got damaged. Thanks, -Original Message- From: [EMAIL PROTECTED] Sent: Wed, 14 Sep 2005

RE: [Asterisk-Users] wait before accepting the call

Why don't you write a couple of lines AGI scripts that will call asterisk command WAIT(5) Thankx -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 27 Sep 2005 13:42:31 +0200 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wait before accepting the call hello!

[Asterisk-Users] ResetCDR with CDR

Hi, I am trying to execute the following asterisk command from one of my AGI script. By providing 'C' flag, I exected CDR would reset. Problem is, CDR was reset but CDR didn't grab destination number (extension) from the Dial command. Well my AGI script was executed after answering a call on a

Re: [Asterisk-Users] cdr_manager.conf

On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote: What is the purpose of cdr_manager.conf? cdr_manager.conf allows you to configure asterisk to send call detail records (cdr) via the Manager API. How I can configure it? to enable CDR via Manager API a cdr_manager.conf looks like

RE: [Asterisk-Users] prepaid application

Hi Scott, Yes, its possible pass 'm' option to Dial command for MusicOnHold If destination is unreachable, you need to get the return value of Dial and from that value you will know whether a call was connected or not. Based on that value you can execute Dial again or not. You can put

RE: [Asterisk-Users] Altering Incoming CallerID

Try, Set(CALLERIDNAME=Innocent Evil) Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 12:48:13 -0500 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Altering Incoming

RE: [Asterisk-Users] Call Recording

What you wanna to do if there have more than 2 parties in the conversation ? !! -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005 19:27:45 +0100 To: asterisk-users@lists.digium.com Subject:

RE: [Asterisk-Users] Write to text file in dialplan

Yes, I have something like this in my extension.conf #include verizon.conf and in verizon.conf file have verizon related dialplan Thanks -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 01 Dec 2005

Re: [Asterisk-Users] Write to text file in dialplan

(echo Call from ${CALLERIDNUM} /tmp/something) but I haven't tested it yet. Can someone confirm this? Innocent Evil wrote: Yes, I have something like this in my extension.conf #include verizon.conf and in verizon.conf file have verizon related dialplan Thanks -- You

[Asterisk-Users] h323 vs oh323

Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com

RE: [Asterisk-Users] h323 vs oh323

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please

RE: [Asterisk-Users] Per Extension Password for Outgoing Routing

You can accomplish password per extention by using an AGI script. model would be, keep extension and password in a table Execute a simple script to authenticate before dial-out You can also accomplish dial-out time from an AGI script. Feel free to ask if you need further help. Thanks, --You

RE: [Asterisk-Users] Call screening script

I have an AGI script but goal was little different. But my script can do this by tweaking little bit. I have a two tables, caller_numbers and statuses every new number is logged in caller_numbers. If that caller call next time, based on the status call can be route to 1. Voicemail 2. Hangup 3.

Re: [Asterisk-Users] CDR MySQL

I was also following this thread. Would anybody please tell, what would be configuration file if mysql is a different machine than asterisk box? Thanks, --You don't have any choice, you already made it before you came here. -Original Message-From: [EMAIL PROTECTED]Sent: Mon, 12 Dec

[Asterisk-Users] caller_id and law

I would like to send auto invitation to members of one of my community organization. I will use a trird party voip provider to make those call out. Question, what caller_id I can pass to that auto invitation message. What does law says? -- You don't have any choice, you already made it before

Re: [Asterisk-Users] caller_id and law

Dec 2005 13:47:18 -0800 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] caller_id and law On Wed, 2005-12-21 at 13:39 -0800, Innocent Evil wrote: I would like to send auto invitation to members of one of my community organization. I will use a trird party voip provider

Re: [Asterisk-Users] caller_id and law

Dec 2005 17:16:11 -0500 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] caller_id and law trixter aka Bret McDanel wrote: On Wed, 2005-12-21 at 13:39 -0800, Innocent Evil wrote: I would like to send auto invitation to members of one of my community organization. I

[Asterisk-Users] wav to g729

hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. Thanks, -- You don't have any

Re: [Asterisk-Users] wav to g729

Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http

Re: [Asterisk-Users] wav to g729

conversion module from redice li ..it is greate! Miklos IPFONE TELEFONIA IP Rua Caio Graco 735 São Paulo SP IPBX - +55 11 3488-3800 http://www.ipfone.com.br [EMAIL PROTECTED] Balbus balbum intellegit - Original Message - From: Innocent Evil [EMAIL PROTECTED] To: Asterisk

Re: [Asterisk-Users] wav to g729

Innocent Evil wrote: I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. redice.krisk.org Do it from Asterisk! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided

Re: [Asterisk-Users] wav to g729

Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. redice.krisk.org Do it from Asterisk! -- Kristian Kielhofner

[Asterisk-Users] AGI variables

When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on

[asterisk-users] not reaching at the destination number I dialed

Hi, From this morning, I am having a wired issue. If I dial a number from my asterisk exteion handset, it end up at different number than I dial and its happening everytime I dialed. I havn't change any thing in my asterisk box. And also, I tried my land phone handset which is not forked to a

RE: [asterisk-users] not reaching at the destination number I dialed

NVM, This is happening for a specific destination numbre only. I am considerting that destination number have a wired problem. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Mon, 31 Jul 2006 08:30:35