[asterisk-users] ISDN data packets

. Or if there is some other command to see these kind of data packets ? Please let me know if I am thinking or looking at something wrong. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University ___ Sign up now for AstriCon

[asterisk-users] ISDN PRI debug in Asterisk

Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command pri intense debug span 1 , does it debug every packet received (control and voice/data packets) ? Thanks -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387

Re: [asterisk-users] ISDN data packets

and I might be looking at things wrongly. Thanks Regards Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Arpit Mehta wrote: I have a ISDN PRI(T1) line coming into my TE110 Asterisk

[asterisk-users] Asterisk Caller ID Info

'Unknown' and not the number of 'Asterisk', is this a feature not supported in Asterisk or is there a problem in my network ? Any hints or suggestions would be really helpful ? Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

[asterisk-users] Setting caller id value on outgoing calls using .call files

as it still displays unknown number. I want set the callerid on the 1.call which is made. exten = _.,4,Set(CALLERID(all)=Joe 911) exten = _.,5,system(cp /var/spool/asterisk/1.call /var/spool/asterisk/outgoing/) Any suggestions how to do that. Thanks a lot. Regards -- Arpit Mehta Graduate Student

Re: [asterisk-users] Setting caller id value on outgoing calls using .call files

Thanks guys. No need to reply. I got my answer from someone. On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: 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

Re: [asterisk-users] Asterisk Caller ID Info

Thanks a lot guys. I got my answer from someone. :) On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Also what are the ways if any to set this DNIS or RDNIS information ? Regards Arpit On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi Asterisk Users, I was wondering why

Re: [asterisk-users] Asterisk Caller ID Info

Also what are the ways if any to set this DNIS or RDNIS information ? Regards Arpit On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: 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

Re: [asterisk-users] Asterisk Caller ID Info

callerid can be set using the variables provided in asterisk, $CALLERID both in .call file (if you are using one) and the extensions.conf file. Regards Arpit On 10/4/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Arpit Mehta wrote: Thanks a lot guys. I got my answer from someone

Re: [asterisk-users] Getting DTMF digits

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 [EMAIL PROTECTED] 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

[asterisk-users] Meet Me sound file

file? How do I go about changing the file with some other sound file ? Thanks a lot Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University ___ --Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Call to an arbitrary outbound number by asterisk

### Goes to the voicemail -- Native bridging Zap/23-1 and Zap/1-1 -- Channel 0/23, span 1 got hangup request -- Hungup 'Zap/1-1' == Spawn extension (incoming, 17689, 1) exited non-zero on 'Zap/23-1' -- Hungup 'Zap/23-1' Regards -- Arpit Mehta Graduate Student Department

Re: [asterisk-users] Call to an arbitrary outbound number by asterisk

Hi, hi, 19173995791 is some number which I want to dial. 212-85- all/most of the numbers in my workplace start with this - so I presume it has got to do something with this. thanks for your suggestions regards arpit On 5/18/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Arpit Mehta [EMAIL

[asterisk-users] Call someone to instantly join conference using MeetMe

requires a person dialing into it and the joining the conference. How could this be done using MeetMe or any other conference application? Any suggestions/hints/links are welcome. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Re: [asterisk-users] Call someone to instantly join conference using MeetMe

Hi Ya that works good. Thanks Arpit On 5/20/07, Kapil Dhawan [EMAIL PROTECTED] wrote: Arpit Use Auto dial. http://www.voip-info.org/wiki-Asterisk+auto-dial+out Create a .call file as mentioned by Dave. Dave Miller wrote: Arpit Mehta wrote on 5/19/07 10:18 PM: I was just wondering how

Re: [asterisk-users] Working softphone for poket PC

___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387

Re: [asterisk-users] How to read SIP debug?

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation provided by Easynews.com

[asterisk-users] Compilation after Source code changes in Asterisk

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

[asterisk-users] Auto Fall Through when kicking users in MeetMe

does auto fall through mean ? Thanks Regards -- Arpit Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

[asterisk-users] Auto Fall Through when kicking users in MeetMe

does auto fall through mean ? Thanks Regards -- Arpit Mehta -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users

Re: [asterisk-users] PRI commands missing...

___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arpit Mehta Graduate Student Department of Computer

[asterisk-users] Get value from linux terminal to dialplan in Asterisk ?

sample code to do so ? Thanks a lot Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Get value from linux terminal to dialplan in Asterisk ?

Thanks !! On 11/1/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Arpit Mehta wrote: 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

[asterisk-users] use dial plan passed arg value in C agi code

by using argc and argv or there is more to be done than that? Thanks Regards -- Arpit Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] use dial plan passed arg value in C agi code

/mailman/listinfo/asterisk-users -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Get IP address of an incoming or outgoing SIP call

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

Re: [asterisk-users] Get IP address of an incoming or outgoing SIP call

* 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

[asterisk-users] REFER mesage extraction using SIP_HEADER

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

Re: [asterisk-users] REFER mesage extraction using SIP_HEADER

options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Print CALLERID in CLI during pri debug

serving the local user (1) I would like to print '1234567890 Message type: CONNECT (7) ... ... ' where 1234567890 is the callerid Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

Re: [asterisk-users] Print CALLERID in CLI during pri debug

Or in other words is there a way to map which message is from which CallerID ? On Dec 6, 2007 6:40 PM, Arpit Mehta [EMAIL PROTECTED] wrote: 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

[asterisk-users] Where does the call go in the dialplan after a call disconnects

Hi all , I would like to do some cleaning up after my call disconnects. For this I need to know where in the dialplan a call goes after it disconnects ? Is there any special place in the dialplan a call goes to when it disconnects ? Thanks Regards -- Arpit Mehta Graduate Student Department

Re: [asterisk-users] Print CALLERID in CLI during pri debug

, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: What don't you tell us what you are ultimately trying to do. You want the callerid next to the connect message in debug output... why? What will that help you to accomplish? On Dec 7, 2007 4:42 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Ok so the call

Re: [asterisk-users] Print CALLERID in CLI during pri debug

messages, like Connect, will reference it. At the same time, the setup will have indicated the caller ID info. Sent from my iPhone On Dec 6, 2007, at 10:28 PM, Arpit Mehta [EMAIL PROTECTED] wrote: Or in other words is there a way to map which message is from which CallerID ? On Dec

[asterisk-users] Check if SIP user is available or not ?

Hi * users, Is there any way to check if a SIP user is currently available or not from the dialplan? By available I mean if a SIP call can be made to that user. Thanks -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998