Re: [asterisk-users] How do you convert your prompts to an asterisk-friendly format?

Hi I wrote this article and at end i shared how to convert files have a look http://younewplanet.com/index.php/articles/2012-articles-2/asterisk-configuration-step-by-step Also i wrote an other article for file conversion you can also check that http://younewplanet.com/index.php/articles/2012-a

Re: [asterisk-users] A worth reading Tutorial for Asterisk Hardware and software configuration

Thank you very much Steve. Thanks for the in detail recommendations. I will make modifications and let you know. On Sat, May 5, 2012 at 9:22 PM, Steve Edwards wrote: > On Sat, 5 May 2012, ABBAS SHAKEEL wrote: > > I wrote this article for making the things easy for newbies Plea

[asterisk-users] A worth reading Tutorial for Asterisk Hardware and software configuration

HI, I wrote this article for making the things easy for newbies Please have a look and let me know if you suggest some thing. I have combined few things to gather i.e 1. Installation Steps 2. Hello World with CDR entry in database 3. Hardware configuration and loop back testing 4. Ast

[asterisk-users] Applet based softphone for Asterisk

Hello Can some body let me know any softphone that is developed using java & can support at least sip protocol. Must be open source and ready to be used. I am trying to accomplish is to integrate it with an applet. some thing like click to call on web page. Sorry if this is not correct place for

Re: [asterisk-users] asking for some help

Hello, I started to work on asterisk 2 years ago. I started from book. I saved it in google docs. You can also start from here . On Thu, Mar 24

Re: [asterisk-users] end a call after a specific time period

exten => _9944NX,1,Answer() exten => _9944NX,2,Noop(GOING FOR THE AGI) exten => _9944NX,3,Noop(XX) exten => _9944NX,4,Noop() exten => _9944NX,5,AGI(//Some script here it works perfectly fine) exten => _9944NX,6,Noop(AG

Re: [asterisk-users] end a call after a specific time period

> I believe absolute timeout will do that. > http://www.voip-info.org/wiki/view/Asterisk+func+timeout > > > > On Sun, Jan 23, 2011 at 2:04 PM, ABBAS SHAKEEL > wrote: >> Hello all, >> I am trying to end a call after a specific time period for that reason i >> hav

Re: [asterisk-users] end a call after a specific time period

in On Sun, Jan 23, 2011 at 10:14 PM, Sherwood McGowan < sherwood.mcgo...@gmail.com> wrote: > On Sun, Jan 23, 2011 at 1:04 PM, ABBAS SHAKEEL > wrote: > > Hello all, > > I am trying to end a call after a specific time period for that reason i > > have tried various opt

[asterisk-users] end a call after a specific time period

Hello all, I am trying to end a call after a specific time period for that reason i have tried various options like using S, L in the dial command. But in vain. Now i am thinking to end the call using the AMI... but i am unable to get the current active channel. . i.e SIP/NT000 when i ask for get

Re: [asterisk-users] E1 configuration

although I don't need the solution personally But would like to request you that instead of posting "forget it" . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 P

Re: [asterisk-users] How to record and playback at the same time

Hello , Record the file and introduce echo this will give you effect of recording and playing at same time ;) On Thu, Jul 29, 2010 at 1:30 PM, Janu Mukherjee wrote: > Hi, > > we are using Asterisk to record and playback. Both services are working > well independently but it seems we can't s

Re: [asterisk-users] How to secure Configuration files

07/07/2010 03:33 PM, Tilghman Lesher wrote: > > On Wednesday 07 July 2010 14:58:05 Kevin P. Fleming wrote: > >> On 07/07/2010 10:52 AM, Tilghman Lesher wrote: > >>> On Wednesday 07 July 2010 05:24:10 A J Stiles wrote: > >>>> On Tuesday 06 Jul 2010, ABBAS S

Re: [asterisk-users] How to secure Configuration files

trade off.. On Wed, Jul 7, 2010 at 2:08 PM, Hans Witvliet wrote: > On Wed, 2010-07-07 at 12:12 +0600, ABBAS SHAKEEL wrote: > > Thanks to Gordon and Paul for kind help. > > > > > > Actually we have a limitation to place the Asterisk server in client > > premises i

Re: [asterisk-users] How to secure Configuration files

thing possible like that ? On Tue, Jul 6, 2010 at 5:21 PM, Gordon Henderson > wrote: > On Tue, 6 Jul 2010, ABBAS SHAKEEL wrote: > > > Hello Community, > > > > I have a question , I have been working with asterisk and developed some > > successful applications.

[asterisk-users] How to secure Configuration files

Hello Community, I have a question , I have been working with asterisk and developed some successful applications. I am facing an issue of security i.e. We deploy servers to client end. Now i dont want the client to see my configuration files (Of course copy and distribute or replicate the logic

Re: [asterisk-users] Asterisk on Ubuntu

If you are using asterisk on your laptop then you dont need dahdi because you are not going to use any hardware on your laptop. How ever for Asterisk start at system start up you will find many scripts One you can do is put "asterisk" at end of /etc/rc.local file On Fri, Jun 4, 2010 at 6:40 P

Re: [asterisk-users] Detect if a Number is up or not

Thanks Loan Indreias ... Nice Idea Thanks Danny Nicholas. Cheers On Tue, Apr 27, 2010 at 7:17 PM, Danny Nicholas wrote: > This is probably a good idea, BUT it is likely that the dialed phone will > never ring (Perhaps that is the desired effect); In my experience it takes > Zap/DAHDI about 2-

Re: [asterisk-users] Detect if a Number is up or not

Thanks Motiejus Jakstsys Thank you for the value able info i will give it a try. 2010/4/26 Motiejus Jakštys > AMI writes event "Ringing...", you can catch it and (via the same AMI) > send a soft hangup request. > > On Mon, Apr 26, 2010 at 12:54 PM, ABBAS SHAKEEL > w

Re: [asterisk-users] Detect if a Number is up or not

l return Dialstatus , if the number dialed is busy or off > now. use this application you can detect a number is busy or not > > in several seconds. i use this method in my dialplan. > > 2010/4/19 ABBAS SHAKEEL : > > Hello Community, > > > > I Want to detect if a cell num

[asterisk-users] Detect if a Number is up or not

Hello Community, I Want to detect if a cell number is ON or OFF... for that matter i can generate call to it using PSTN lines (configured with asterisk). The problem is that i only want to see if the cell number can receive a ring or not. If ring is recieved at called number end then mark it as

Re: [asterisk-users] Strange Centos Problem with Dahdi installation

t; > If your processor type is *i686*, you just need to install > kernel-PAE-devel-2.6.18-128.el5.i686.rpm like following: > > rpm -ihv > ftp://ftp.pbone.net/mirror/ftp.centos.org/5.3/os/i386/CentOS/kernel-PAE-devel-2.6.18-128.el5 > .*i686*.rpm > > BR, Alexey > > 2010/4/

Re: [asterisk-users] Strange Centos Problem with Dahdi installation

system... This issue has ruined my continuous 36 hours with out sleep One thing i have in mind is to install ubuntu instead of centos to get rid of this issue. I may try this on Monday. On Fri, Apr 2, 2010 at 7:45 PM, Warren Selby wrote: > On Friday, April 2, 2010, ABBAS SHAKEEL >

Re: [asterisk-users] Strange Centos Problem with Dahdi installation

thanks Tim for the reply As i said I have tried to install the kernel-devel package. But after successfull installlation when i reboot conflicts occurs between the kernels as a result i am unable to boot .. On Fri, Apr 2, 2010 at 5:29 PM, Tim Nelson wrote: > On 4/2/2010 1:50 AM, ABBAS SHAK

[asterisk-users] Strange Centos Problem with Dahdi installation

Hello Community, I have installed Dahdi on Centos on many system and succesfully used that.. But today i have bad luck... This is the error that i am facing [r...@localhost dahdi-linux-complete-2.2.1+2.2.1]# make all make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.

Re: [asterisk-users] Is there any Diguim distributor in Lahore

HEllo try this http://www.voip-info.org/wiki/view/Digium On Fri, Mar 26, 2010 at 3:29 PM, Faheem wrote: > Hey, is there any Diguim distributor in Lahore,Pakistan? I need to buy > X100P. > > Muhammad Faheem > > > > > -- > _ > --

Re: [asterisk-users] pstn calls not picked up

Hello, Please Confirm if the dahdi/Zaptel service is running . check your channels status. On Wed, Mar 24, 2010 at 9:29 AM, Balu Raman wrote: > I have a PSTN line coming into FXO port 4 on a TDM400P. Incoming calls > are not being picked up. I don't find anything unusual in asterisk > log. I

Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

Thanks alot for the value able information. My client is not sure about the requirements as he reaches a final decision then i can move forward to start working on it. Thanks for the info. On Fri, Mar 19, 2010 at 6:30 PM, Jonathan Addleman wrote: > Philipp von Klitzing wrote: > >> I would like t

Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

Thanks I will look into it. On Fri, Mar 19, 2010 at 2:26 AM, Philipp von Klitzing < klitz...@pool.informatik.rwth-aachen.de> wrote: > > I would like to know if any one have experience with live audio > > streaming like 1. Streaming from an online resource > > Look at app_ices and icecast. > http:

[asterisk-users] Live Audio Streaming- From Aux interface-Online resource

Hello all, I would like to know if any one have experience with live audio streaming like 1. Streaming from an online resource 2. Streaming from sound card AUX interface.. What i want to accomplish is that on receiving a callers call i play back a live audio stream or stream from sound card AUX

Re: [asterisk-users] How to detect a PSTN telephone is busy or not?

Hello, Please have a look to DIALSTATUS variable. here http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS I hope it helps On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun wrote: > hi,all > > one problem confuse me the

Re: [asterisk-users] Asterisk Cepstral TTS

uys... > I already have Cepstral installed I guess I just need to figure out > where in the .call file and format to call cepstral and then the txt > for the message. Thanks again for all of your help! > > > > On Fri, Feb 12, 2010 at 11:50 AM, ABBAS SHAKEEL > wrote: >

Re: [asterisk-users] Asterisk Cepstral TTS

hello , First you check out http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Once you are done with auto dial out then look for Cepstral TTS. http://www.google.com.pk/search?hl=en&safe=active&q=asterisk+with+cepstral&btnG=Search&meta=&aq=f&oq= One more thing that there are oth

Re: [asterisk-users] SS7 and Asterisk

Please some one shed some light on it.. On Thu, Feb 4, 2010 at 6:48 PM, ABBAS SHAKEEL wrote: > Hello All, > Please let me know Answers to the following questions .Backgroud. > > 1. Which one is better to use libss7 or chan_ss7. Today first time i come > to know about it ... littl

[asterisk-users] SS7 and Asterisk

Hello All, Please let me know Answers to the following questions .Backgroud. 1. Which one is better to use libss7 or chan_ss7. Today first time i come to know about it ... little bit i googled but need experts comment on it. 2. I have perpared my server ie installed Asterisk , Configured TE420P &

Re: [asterisk-users] You won't help me anymore?

why don't you post your question On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi wrote: > > > On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra wrote: > >> Sunday, January 10, 2010, 11:24:22 AM, hadi wrote: >> >> > You are not willing to help me anymore ? >> >> Why do you think this? >> >> -- >> Be

Re: [asterisk-users] Fastagi-mapping problem

You can try this [agi_test] exten => 123,1,Answer(); exten => 123,n,noop(${CALLERID(num)}) exten => 123,n,set(IP_FOR_AGI=192.168.127.58) exten =>123,n,Agi(agi://${IP_FOR_AGI}/com.package.ClassName) On Wed, Jan 6, 2010 at 2:28 PM, ABBAS SHAKEEL wrote: > Hi, > > You can dir

Re: [asterisk-users] Fastagi-mapping problem

Hi, You can directly call that class like AGI(com.abc.cde.Hello) . Hello is class name. Hope this helps On Wed, Jan 6, 2010 at 2:16 PM, ahmed magdy wrote: > Hello > > I am new in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started > the first example Hello AGI in this tutorial > http:

Re: [asterisk-users] Asterisk depend on postgresql files?

Can some body shed some light on this please On Mon, Dec 21, 2009 at 6:41 PM, ABBAS SHAKEEL wrote: > Hello > > when compiling asterisk with Postgresql we need to specify directory where > the postgresql is installed. > It uses some files from bin folder of postgresql (I am not

Re: [asterisk-users] Every one Busy Problem

left to explain Cheers On Tue, Dec 22, 2009 at 2:21 PM, Tzafrir Cohen wrote: > On Tue, Dec 22, 2009 at 11:41:35AM +0500, ABBAS SHAKEEL wrote: > > Hello > > > > When ever i try to use Dial DAHDI / SIP i get the following warning and > > nothing happens and Asterisk moves

[asterisk-users] Every one Busy Problem

Hello When ever i try to use Dial DAHDI / SIP i get the following warning and nothing happens and Asterisk moves to next instruction Even i know that channel is free no one else is using it [Dec 22 12:43:39] WARNING[11915]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI'

[asterisk-users] Asterisk depend on postgresql files?

Hello when compiling asterisk with Postgresql we need to specify directory where the postgresql is installed. It uses some files from bin folder of postgresql (I am not a developer of asterisk but a user ). I need to know once asterisk is ready to use(ie compiled and installed ). Do it still refe

Re: [asterisk-users] TE420P configuration - Loopback

Dear Daniel, I am keeping this on list so that it can be help ful to others as well. >>>They ask me to return back to them the data packets they send me through. In my opinion They are sending you data packets on some channel and you need to reply back from another channel ie se

Re: [asterisk-users] TE420P configuration - Loopback

No Problem you are Welcome. But please try to be on the list Loop back connector can be used to test your card . For loop back connector you have to make a loopback cable. that you will plug in card. its image is here

Re: [asterisk-users] Example to handle incoming calls without callerid at home?

James you are right. let me add one more line exten => s,n,GotoIf($[${CALLERID(num)}=""]?nocid,s,1) On Sun, Dec 6, 2009 at 3:18 PM, James Stocks wrote: > On 6 Dec 2009, at 08:56, Remco Barendse wrote: > > > I am using asterisk 1.6 at home and would like to send incoming calls > > without caller

Re: [asterisk-users] Dial application with M option

Aah the Problem was i am working on 1.4 and in my mind and logic i was writing code for 1.6. The example works perfect On Thu, Dec 3, 2009 at 3:00 PM, ABBAS SHAKEEL wrote: > Any one have success with Dial M option, Can some one provide an example? > > > On Thu, Dec 3, 2009 at 12:

Re: [asterisk-users] Dial application with M option

Any one have success with Dial M option, Can some one provide an example? On Thu, Dec 3, 2009 at 12:57 PM, ABBAS SHAKEEL wrote: > Hello, > > What i am trying to do is . Dail a number and ask if you wana talk to > XXX press 1 and if you dont wana talk press any other key. > For

[asterisk-users] Dial application with M option

Hello, What i am trying to do is . Dail a number and ask if you wana talk to XXX press 1 and if you dont wana talk press any other key. For this purpose i am using this link . *I am using this option :- * *M(**x**)*: Executes the macro (x)

Re: [asterisk-users] Asterisk Configuration with Sphinx speech engine

Hello I also tried it in begining but cant give time to it. So no success. you can try this link http://www.voip-info.org/wiki/view/Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php hope this helps On Tue, Dec 1, 2009 at 12:16 PM, Riz

Re: [asterisk-users] ChanIsAvail querry

Thanks Michiel & Covici @ Michiel i will try the script @Covice yes it is a DAHDI channel On Wed, Nov 25, 2009 at 8:12 PM, wrote: > I wonder if this is related to my problem where the channel returns with > a status of BUSY even if it is on hook -- this is a dahdi channel. > &g

Re: [asterisk-users] ChanIsAvail querry

Dan I have reverted to 1.4.27 but got no success. Same behaviour Do anyone has any success with it ? On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL wrote: > Thanks Michiel and Dan > > @ Michiel i have checked the variables but they dont contain any value. > @Dan I am using 1.6.1.2

Re: [asterisk-users] ChanIsAvail querry

> https://issues.asterisk.org/view.php?id=14426 – link to the issue > > > > Hope that helps. > > Dan Journo > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *ABBAS SHAKEEL > *Sent

[asterisk-users] ChanIsAvail querry

Hello We need to know if a channel is not in use and can be used to dial a number etc.. I have tried the ChanIsAvail function with different parameters. ie ChanIsAvail(DAHDI/1&DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc no matter the channel is busy or not it always return 0 . Please suggest FYI Ch

Re: [asterisk-users] Get the extension dailed

c] > > exten => _.,1,Answer > exten => _.,n,NoOp(${EXTEN}) > > ABBAS SHAKEEL wrote: > > > Thanks Alex, > > > > suppose this is the context > > > > [abc] > > exten => s,1,Answer(); > > exten => s,n,Noop(${EXTEN}); > >

Re: [asterisk-users] Get the extension dailed

AHDI/2-1", "") But i need the number that the user is dialing not the "s"; On Mon, Nov 23, 2009 at 3:24 PM, Alex Balashov wrote: > ABBAS SHAKEEL wrote: > > > I am facing problem while getting the number dialed. if the user is > > using SIP phone then w

[asterisk-users] Get the extension dailed

Hello When a user makes a call to an Asterisk system, He dials a number . We need to know that dialed number. We can get the dialed number by using CALLERID(dnid) and we can get the CLI information using CALLERID(num). I am facing problem while getting the number dialed. if the user is using SIP

Re: [asterisk-users] Minimum hardware requirements for 10 concurrent calls?

Hello Veselin Please try this http://www.google.com.pk/search?hl=en&safe=active&ei=PnUGS9mhLMvanAfUuozGCw&sa=X&oi=spell&resnum=0&ct=result&cd=1&ved=0CBIQBSgA&q=Minimum+hardware+requirements+for+Asterisk&spell=1 On Fri, Nov 20, 2009 at 3:39 PM, Veselin K wrote: > Any advise? > > Thank you. > > Ve

Re: [asterisk-users] Saving CDR on Different Databases

ow wich row I have to copy. > > regards > > Mickael > > 2009/11/18 ABBAS SHAKEEL > >> Hello >> If we need to save CDRs on different databases for same Asterisk server >> ie >> suppose for context [abcd] save to local:5432:abcd >> and for context

[asterisk-users] Saving CDR on Different Databases

Hello If we need to save CDRs on different databases for same Asterisk server ie suppose for context [abcd] save to local:5432:abcd and for context [wxyz] save to local:5432:wxyz Can we manage it ? or we need to do some thing in AGI -- Kind Regards Shakeel Abbas

Re: [asterisk-users] GSM and Wav format

Thanks Alot all. Specially Tim It seems to be really good. I will check it in detail On Sun, Nov 15, 2009 at 3:44 PM, Tim Panton wrote: > > On 2 Nov 2009, at 12:11, ABBAS SHAKEEL wrote: > > > Hello, > > Let me explain a scenario > > There are different Asterisk

Re: [asterisk-users] FW: hi Dan

I cant stop laughing lolz Any how we must not reply in private but ask to post on list only. Lets make him able to achieve his objective through the list. Cheers On Sat, Nov 14, 2009 at 11:56 PM, Alex Balashov wrote: > I don't get it. I just replied helpfully to Mr. "opensourcesolutions" > on

Re: [asterisk-users] FW: hi Dan

This is happening here also :( On Fri, Nov 13, 2009 at 9:02 PM, Cary Fitch wrote: > Sorry, I can't resist. > > How do I join the Mail List Nazi Corp? Do I have to be invited, or can I > just self appoint myself? Asking neophyte questions are objected to by > some, top posting by those who blast

Re: [asterisk-users] soft phone (X-lite) not able to register with asterisk

Please stay on list because if some one other face similar problem he can get help by googling list. IN domain name u have to specify ASterisk server IP in XLITE. On Thu, Nov 12, 2009 at 11:36 AM, ABBAS SHAKEEL wrote: > I have replied you already . Please look into it > > On Th

Re: [asterisk-users] soft phone (X-lite) not able to register with asterisk

I have replied you already . Please look into it On Thu, Nov 12, 2009 at 11:31 AM, wrote: > > hi all, > i had installed asterisk on Centos 5.3, sip.conf and extentions.conf are > *vi /etc/asterisk/sip.conf* > [general] > port = 5060 > bindaddr = 192.168.1.2 (asterisk server ip addr) > conte

Re: [asterisk-users] softphones (x_lite) not able to register with asterisk server

You may be doing some thing wrong with Configuration of Softphone. Please take a tutorial .. Google is a good friend. I suggest you to use X-lite softphone. On Thu, Nov 12, 2009 at 11:25 AM, ABBAS SHAKEEL wrote: > > Hello. > > I see this post many times. I have written this for

Re: [asterisk-users] softphones (x_lite) not able to register with asterisk server

Hello. I see this post many times. I have written this for you to get a start. This is sip.conf [general] context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060; UDP Port to bind to (SIP standard port is 5060)

Re: [asterisk-users] Call audio leaking between calls

Aslamoalikum Ishfaq Can you check this with asterisk 1.6.X ? On Tue, Nov 10, 2009 at 2:45 PM, Ishfaq Malik wrote: > Hi > > Has anyone ever had experience of phones on the same office network > being able to hear other concurrent call's audio whilst on calls of > their own? We're getting this f

Re: [asterisk-users] GSM and Wav format

After conversion from .wav to .mp3 the size remains almost the same. On Mon, Nov 2, 2009 at 5:46 PM, ABBAS SHAKEEL wrote: > Thanks Patrick. > > First: I dont do that intentionally. > > Thanks for suggestion. Let me investigate it. > > > On Mon, Nov 2, 2009 at 5:34 P

Re: [asterisk-users] GSM and Wav format

> You can convert wav to mp3 on the recording server and then send it to > the central system. > > Bye, > Patrick > > On Mon, Nov 2, 2009 at 1:11 PM, ABBAS SHAKEEL > wrote: > > > > Hello, > > > > Let me explain a scenario > > > >

[asterisk-users] GSM and Wav format

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to t

Re: [asterisk-users] Asterisk Server with Panasonic PBX

ri, Oct 30, 2009 at 3:41 AM, C F wrote: > On Wed, Oct 28, 2009 at 10:57 PM, ABBAS SHAKEEL > wrote: > > C F thankyou very much. > > > > when i make a call to Asterisk server recieves and works fine. But as to > > make external calls we have to press nine so supposed a

Re: [asterisk-users] Asterisk Server with Panasonic PBX

the TDA depending on how you chose to > connect them both. > > On a side note, may I ask why you are integrating asterisk with the > TDA? What is the functionality you plan on gaining? > > On Wed, Oct 28, 2009 at 4:50 AM, ABBAS SHAKEEL > wrote: > > Hello > > I ha

Re: [asterisk-users] Syncronizing files on different Asterisk servers

Thanks all Robin Drop Box looks cool but I have developed my own code in JAVA that will use Sockets to syncronize files across different servers. Thanks Arjan for the link. @ li...@torrenga.com yeah i do have considered but finally developed my own code for sysncronization. thanks :) if Any One

[asterisk-users] Asterisk Server with Panasonic PBX

Hello I have a scenerio to integrate an Existing Panasonic PBX with a new PBX that will be Asterisk system. I know that Asterisk can be integrated with existing Panasonic TDA 100 PBX to recieve calls (ie PSTN lines to Panasonic PBX and out lines of Panasaonic to in of Asterisk PBX). -->But i am i

Re: [asterisk-users] Recording management for IVR

hello AGI is a good option to handle such complexities On Fri, Oct 23, 2009 at 6:33 PM, Mail list wrote: > Hello everyone. I have a client with specific requirement, here's the > scenario: > > Call comes in > Ivr menu, press 1 for new record 2 for existing 3 for operation blabla.. > on pressing

Re: [asterisk-users] Syncronizing files on different Asterisk servers

I am considering these options On Wed, Oct 21, 2009 at 10:08 AM, Joseph wrote: > On 10/20/09 17:24, ABBAS SHAKEEL wrote: > >Hello > >I need some advice regarding the Asterisk server that are located at > >different locations. > > > >Three asterisk servers ar

Re: [asterisk-users] Syncronizing files on different Asterisk servers

t 2009, ABBAS SHAKEEL wrote: > > > Hello > > I need some advice regarding the Asterisk server that are located at > > different locations. > > > > Three asterisk servers are here each at different location. Suppose A,B,C > be > > the three servers respectivel

[asterisk-users] Syncronizing files on different Asterisk servers

Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR ser

Re: [asterisk-users] Check if a variable is set

gt; exten => s,2,ExecIf(EXISTS(${ARG3})=1 & ${ARG3}=1| > > I see this in the console > Executing [...@macro-extcall:2] ExecIf("SIP/PACK501-08222428", "EXISTS()=1 > & =1| > > As I didn't pass a third argument. > > Essentially, what I'm try

Re: [asterisk-users] Check if a variable is set

Sorry its macro I called it a function. This link will be helpful to you http://www.voip-info.org/wiki/index.php?page=Asterisk+variables <http://www.voip-info.org/wiki/index.php?page=Asterisk+variables> On Fri, Oct 16, 2009 at 3:13 PM, ABBAS SHAKEEL wrote: > If you want to check in Con

Re: [asterisk-users] Check if a variable is set

If you want to check in Console then NOOP can be used .if in case of function call you can check its length if there exists some thing On Fri, Oct 16, 2009 at 3:04 PM, Ishfaq Malik wrote: > Hi > > Is there any way to check if a variable is set in asterisk? I've had a > look around and can't find

[asterisk-users] Trunk and Pstn line

Hello Please let me know can we call normal PSTN lines as trunk lines?? As a normal pstn line used in home . One More thing that If i need ten PSTN lines on one Server then which Digium card is suitable. I am confused with TDM800P as it say it accepts a trunk line? -- Best Regards Shakeel Abba

Re: [asterisk-users] Softphone in Web

Thanks. But Can i enhance it in such away that it can make calls to asterisk as part of a web application ?? user can call from webapplication i think mozphone is a plugin for mozilla... On Thu, Oct 1, 2009 at 2:00 PM, Administrator TOOTAI wrote: > ABBAS SHAKEEL a écrit : > &

[asterisk-users] Softphone in Web

Hello I am thinking to develop a softphone that is integrated into web.(in form of APPLET or some thing else) Ie a user with with just a PC with Net Browser(fire fox etc) Installed can make call.. Is there some thing developed before like this that is open source ?? -- Best Regards Shakeel

Re: [asterisk-users] VOIP solutions

wrote: > > Don't put a SIP server behind destination NAT. Just don't. > > Why not? Mind to explain? > > > > > ABBAS SHAKEEL wrote: > > > >> Sorry My Question was not very clear. > >> > >> Asterisk System that is placed some where on lo

Re: [asterisk-users] VOIP solutions

Thanks Alex By just avoiding this will solve this problem? On Sat, Sep 26, 2009 at 9:47 PM, Alex Balashov wrote: > Don't put a SIP server behind destination NAT. Just don't. > > ABBAS SHAKEEL wrote: > > > Sorry My Question was not very clear. > > > > Aste

[asterisk-users] VOIP solutions

Hello I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configur

Re: [asterisk-users] Inquiry:How to convert *.wav files ?

Hello Hadi While playing files extension is not specified. Remove the extension and Enjoy On Sat, Sep 26, 2009 at 3:13 PM, ravi kumar wrote: > Use > Audocity Software > > Ravindra kumar > > > On Sat, Sep 26, 2009 at 11:14 AM, hadi motamedi wrote: > >> Dear All >> Can you please do me favor and l

Re: [asterisk-users] VOIP solutions

this sip phone. As Asterisk Server is also behind NAT. SIP phone is also in any other network. How can I make them communicate. As in LAN i can easily by giving asterisk server IP. On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen wrote: > Abbas Shakeel wrote: > > I Recently complet

Re: [asterisk-users] Inquiry:How to convert *.wav files ?

A good way is to give try On Sat, Sep 26, 2009 at 11:41 AM, ABBAS SHAKEEL wrote: > yeah it can :) > > > On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi wrote: > >> Thank you for your reply . Excuse me , you mean the Asterisk can play SLN >> files ? Can you please c

Re: [asterisk-users] Inquiry:How to convert *.wav files ?

yeah it can :) On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi wrote: > Thank you for your reply . Excuse me , you mean the Asterisk can play SLN > files ? Can you please confirm ? > > > > On Sat, Sep 26, 2009 at 6:57 AM, ABBAS SHAKEEL < > shakeel.abbas@gmail.com

Re: [asterisk-users] Inquiry:How to convert *.wav files ?

Hello Hadi In beginning i also face this problem . I solved it by converting to SLN format. You also try to convert it to sln format. this link might help you http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk

[asterisk-users] Dynamically Move caller to different dial plan

Hello all Is it possible for Asterisk (it can be Asterisk Manager) to A caller who is flowing in any dail plan (Say Dialplan A) ...On a particular event ( can be generated by any other caller) pick him up to any other dial plan(say Dial plan B) without his(/her) wish. .. and after taking some

Re: [asterisk-users] All the four lights blinking

},60,R) On Sat, Sep 12, 2009 at 3:23 AM, Christian Victor wrote: > 2009/9/11 ABBAS SHAKEEL > >> Thanks you very much Kevin.I will try it by connecting one end of >> Ethernet cable to one slot and other to second slot . Configuring one >> as pri_net and

Re: [asterisk-users] All the four lights blinking

Thanks you very much Kevin.I will try it by connecting one end of Ethernet cable to one slot and other to second slot . Configuring one as pri_net and the other as pri_cpe. I will provide you feed on monday either i succed or not On Fri, Sep 11, 2009 at 4:23 PM, Kevin P. Fleming wrote: > AB

[asterisk-users] A Strange Exception/Notice

Hello All I am facing a strange exception/Notice while running an Asterisk box. Let me explain in detail I have asterisk box 1.6.1.2 centos 5.2 I have connected TE420P to the box. then connected a loopback connector( http://i580.photobucket.com/albums/ss246/shakeelabbas/loop_back.jpg). The light

Re: [asterisk-users] All the four lights blinking

too. But I cant generate calls using the loop back connector and get the following error *CLI> [Sep 9 14:42:55] NOTICE[9981]: channel.c:3749 __ast_request_and_dial: Unable to request channel DAHDI/1/123 THANKS On Wed, Sep 9, 2009 at 12:47 PM, ABBAS SHAKEEL wrote: > HelloI have t

[asterisk-users] All the four lights blinking

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage

Re: [asterisk-users] E1 line simulation for Asterisk

> > > > *De:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *En nombre de *ABBAS SHAKEEL > *Enviado el:* Lunes, 07 de Septiembre de 2009 11:33 p.m. > *Para:* Asterisk Users Mailing List - Non-Commercial Discussion > *Asunto:* Re:

Re: [asterisk-users] TE420P configuration

/etc/dahdi/system.conf file is auto generated do we need to change in this file as we do for zaptel ? Any working examples ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register

Re: [asterisk-users] E1 line simulation for Asterisk

Hello I have the loop back connector and TE420P card but i dont know how to configure that. Please let me know of any help. I am facing the problem in configuration of channels. i have make changes in chan_dahdi [r...@te420 etc]# dahdi_hardware pci::04:08.0 wct4xxp+ d161:0420 Wildcard

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