similar to: Channels ID / Soft Hang Up

hi all, how to set the caller id facility for the TDM400p card. Please help me thanks, sandeep.s ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Tuesday, January 15, 2008 3:09 PM Subject: asterisk-users Digest, Vol 42, Issue 51 > Send asterisk-users mailing list submissions to > asterisk-users at

Hi, I use asterisk 1.4.11 version for making outbound calls. Running it on linux(fedora core 7) machine. Recently purchased the g729 codec, got it registered with my asterisk box. I have two queries for you to help me. 1. How do i know when an outbound call is placed that it makes use of the g729 codec. when i use the command "show g729" i get the following: 0/0 encoders/decoders of

Hi, I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift text to audio. My functionality is as for example i make this text to audio deliver the person called. Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 "Press 1 to confirm. Press 3 to cancel." extension.conf dialplan: [dialout] exten =>

Hi, I have been working with asterisk to make ivr calls (outbound and inbound). I have the functionality - Read(variable|file_name) used in my dialplan. Now i need to pass the variable to my ruby file to compare the data entered with the database (mysql). How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Any one has any clues on

Hi, I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try making a simple incoming call using xlite softphone. I get the following message when i try calling to the number. *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No compatible codecs, not accepting this offer! Which codec is it talking abt here. How can i resolve this. My dialplan is as

Hi, Was just wondering about the bug fix in asterisk-1.6 Beta Release.. Its been said that "Added an 'n' option to SpeechBackground to request that the channel not get answered". Can anyone brief me about what exactly this bug are they talking about. We had issues with Background syntax working in macro. Is this the bug fix they are talking about. Regards, Naveen.Palani

Hi, I have been using asterisk-1.4.17 version. Have a SIP registration from bandtel sip providers. Use DID numbers for the incoming calls which works fine when i dont use any peer setting in my sip.conf file. But when i use a peer and make calls thru the DID number it doesn't reach asterisk at all. Doesnt give me any errors as well. peer in my sip conf is as given below: [proxy2_bandtel]

Hi, I have been working with Asterisk for the ivr functionalities in the past. I am interested to implement the Jabber - Gtalk in asterisk. For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command "make menuselect" the channel driver "chan_gtalk" shows xxx

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi Steve; Currently with Dahdi, for the below configuration that we were using it in the rc.local, how it will be? I think there should be something new to be used instead of ztcfg, what it is? And what about other lines? They need to be changed? touch /var/lock/subsys/local /sbin/modprobe wctdm /sbin/ztcfg -vv /usr/sbin/fxotune -s /usr/sbin/safe_asterisk Regards Bilal --- On Thu, 5/1/08,

Hi, I am new user on asterisk (for that matter linux) and i have lot of embedded programming experience. We have a new project from our client, to design a box that takes the telelphone line as input and route the line to the respective user with different ring tones. The box should be programmed by the users with buttons. Features. 1. I should be able to store some .wav files for different

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

I'm using SixTel as a test (Opened account w/ $10) and am happy with them so far... In their basic service package, they don't charge a monthly fee, and it's outbound only, and you get charged for every minute. I paid for a DID, which is $1.50 or so per month, and it lets me receive inbound calls, which I also pay for by the minute.. I don't mind this for a service like this

n active project, than a dead one. Otherwise who is going to patch vulnerab= ilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What

with an active project, than a dead one. Otherwise who is going to patch vulnerabilities? Not me. I'm not a software developer. - Doug Mortensen Network Consultant Impala Networks P: 505.327.7300 . From: Steve Totaro [mailto:stotaro at totarotechnologies.com]=20 Sent: Thursday, March 24, 2011 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

Ok.. will be there... -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Thursday, October 04, 2007 12:50 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 39, Issue 12 Send asterisk-users mailing list submissions to

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a

On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher > <tilghman at mail.jeffandtilghman.com> wrote: >> It is completely illegal in any country that recognizes patents. > > You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we

On Sun, Jun 7, 2015 at 10:05 AM, Luca Bertoncello <lucabert at lucabert.de> wrote: > Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > > Are you using the wifi on on the cellphone? The peer IP is showing as >> 192.168.200.3 which is not a routable address. Unless things have >> changed, >> double NAT configurations do not work. >> >