similar to: ISDN PRI debug in Asterisk

Hi all, I was wondering if it is possible to print the callerid value in the CLI when doing 'pri debug span 1' For example > Call Ref: len= 2 (reference 2707/0xA93) (Terminator) > Message type: CONNECT (7) > [18 03 a9 83 97] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 > ChanSel: Reserved >

Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten

Hi Asterisk Users, I was wondering why a call that is received from Asterisk shows a caller ID 'Unknown' . So here is the scenario, 'A' calls 'Asterisk'. 'A' joins a conference on 'Asterisk'. 'Asterisk' calls 'B'. 'B' gets joined to the same conference also. 'B' somehow receives the caller ID 'Unknown' and not the

Hi * Users, What is the way from the dial-plan to get the IP address of an incoming or outgoing SIP call? I can see the IP address of the SIP call using 'sip show peers' from the CLI. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Hello * users, I know that passing variable in the AGI script is by exten => _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being passed and simple_c_prgm is C code Now how will I receive these variables within C code ? Is it by the same way arguments are passed in command line to C by using argc and argv or there is more to be done than that? Thanks Regards -- Arpit Mehta

Hi, I have a strange problem. I have a TE110p digium card. I want to dial 19173995791 when any incoming call comes in. What is happening is that when I dial 19173-995791. Asterisk picks up the first 5 digits assuming it is the extension and appends 212-85 (here in the university most numbers start with this) in front . Therefore I get connected to some random number 212-85-(19173) (where the

Hi all, My scenario is such that I have three users connected to a conference. CLI> meetme list 1234 User #: 01 9176502096 <no name> Channel: Zap/23-1 (unmonitored)00:00:32 User #: 02 john john Channel: SIP/john-b7800468 (unmonitored) 00:00:28 User #: 03 6463875998 <no name> Channel: Zap/22-1 (unmonitored)00:00:19 3 users in that

Hi, I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or

Hello Asterisk Users, I wanted to know a simple way in which I could read some file from a console (say by using system command) and based on that either return true or false back to dialplan. Is there any built in command in Asterisk for that ? What are the options do I have ? Are there any sample code to do so ? Thanks a lot Regards -- Arpit Mehta Graduate Student Department of Computer

I forgot to add that this is a T1 ISDN PRI line on which I am sending the DTMF digits. Regards Arpit On 10/5/07, Arpit Mehta <arpitm at gmail.com> wrote: > Hi, > > Is there any way to get the DTMF digit preferably in the > extensions.conf . The dtmf digits would be entered by the user > like"1234567890P1234#" . It doesnt matter whether to put 'P' or

Hi * users, I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Hi all, i need to study the SIP protocol. can anybody tell me about any ebook which could halp me understand the sip protocol, architecture, and how to read and understand the sip signalling when i use "sip debug" in asterisk? -- Rizwan Hisham Software Engineer AXVOICE Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone explain to me the sql query done in the last step : http://pastie.org/402200 -- Arpit Jain --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to rubyonrails-talk-/JYPxA39Uh5TLH3MbocFFw@public.gmane.org To unsubscribe from this group,

Hi all. While installing gems i am getting an error: ERROR: Error installing gem xml-simple-1.0.11.gem[.gem]: install directory thname:e:/ruby/lib/ruby/gems/1.8/gems/xml-simple-1.0.11> not absolute Ruby Version = 1.8.4 Rails Version = 1.2.5 Gems Version = 0.9.4 can anybody provide any pointers??? Thanks Arpit.

hi, This might be the most obvious thing to you. I need to change some parts of the source code of Asterisk. I was wondering if we have to compile the whole source code again everytime using the commands (which i think might take some time to compile again) cd /usr/src/asterisk-version make make install or is there a faster and better way to do things Thanks a lot for all the help i have

Hello, I am trying to get the trace by running my sample program using llvm. I was referring following link for lli: http://llvm.org/releases/1.0/docs/CommandGuide/lli.html after that I installed llvm and llvm-gcc. Here is the version detail: tushar at tushar-mehta:~/NUS/os_project/tryout$ lli -version Low Level Virtual Machine (http://llvm.org/): llvm version 2.8 (Ubuntu 2.8-0ubuntu1)

Sir, I am Adit Mehta first year student of DA-IICT. I know C/C++, python, ruby, PHP. I am new to llvm. I am interested in contributing to llvm. Can you guide me from where should I start? Thanking you for anticipation, Adit Mehta -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives

Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100127/b13d6a62/attachment.htm

I have not been able to find any programs for running vector autoregressions with R. I am interested in running Bayesian VARs and also running VARs that run all combinations of variables in the vector. Is anyone currently developing this? -Nirav Mehta