Displaying 20 results from an estimated 900 matches similar to: "Get IP address of an incoming or outgoing SIP call"
2007 Dec 06
2
Print CALLERID in CLI during "pri debug "
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
>
2007 Oct 04
5
Setting caller id value on outgoing calls using .call files
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
2007 Oct 04
1
Asterisk Caller ID Info
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
2007 May 17
2
Call to an arbitrary outbound number by asterisk
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
2007 Sep 18
2
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: 1-646-387-5998
2007 Nov 02
3
use dial plan passed arg value in C agi code
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
2007 Jul 08
2
Auto Fall Through when kicking users in MeetMe
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
2007 May 19
1
Call someone to instantly join conference using MeetMe
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
2006 Feb 09
2
IP Authorization
You can use the following:
switch3*CLI> show function SIPCHANINFO
switch3*CLI>
-= Info about function 'SIPCHANINFO' =-
[Syntax]
SIPCHANINFO(item)
[Synopsis]
Gets the specified SIP parameter from the current channel
[Description]
Valid items are:
- peerip The IP address of the peer.
- recvip The source IP address of the peer.
- from
2007 Nov 02
1
Get value from linux terminal to dialplan in Asterisk ?
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
2008 Mar 25
1
How to obtain SIPCHANINFO variables within custom application?
Hello,
How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough
variables in (within) my custom Asterisk application?
I can't use chan_sip.c internal structures (such as sip_pvt) in my custom
application, because there's no chan_sip.h and I can't include it into my
application (maybe there's other way?).
I can do like this:
exten =>
2007 Oct 07
0
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 <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
2007 May 31
2
How to read SIP debug?
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.
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2012 Oct 05
3
How to log caller IP address in the CDR?
Hello
We had this situation:
Some bot-net did try to guess SIP logins and finally succeeded. The Asterisk
Server was abused to call a large number of expensive destinations.
It is clear that the sip logins have been passed to various persons (probably
posted on a forum somewhere inviting to do 'free calls').
Right after the affected password was changed, the message log shows which
2007 Oct 19
2
Howto get origin IP address from SIP call reliably
Hi,
incoming SIP calls have a channel name in the form of:
SIP/<ip-adresss-of-peer>-<handle>
This is a way to get fetch the IP address of the remote side
of a SIP call - in most cases.
However, sometimes, instead of the IP address, there is a host
name in the channel name. I assume, this value in the channel name
is not the real IP address, but just a field filled in by the
remote
2009 Feb 21
1
VoIP Information in CDRs
Hi,
I am trying to find a way to add the following info in CDRs (with
asterisk 1.4.23.1):
1. Codec used
2. RTP QoS statistics
3. RTP IP of remote host
4. For answered calls, the peer that requested to end the conversation
I have managed to get 1 and 2 for the caller, like that:
exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}
2009 Feb 27
5
Polymorphic association..explain the extra query ?
Can anyone explain to me the sql query done in the last step :
http://pastie.org/402200
--
Arpit Jain
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2006 Apr 05
15
How to restrict simultaneous phone registrations
Hello all,
I am looking for a way to restrict users from logging in two separate
phones with the same authorization name/password at the same time.
Meaning, I only want users to be able to place a call from one phone in
one location, but have the ability to move from computer to computer.
Has anyone found any sort of solution for this type scenario?
Thanks,
Bryan Mahin
Please visit us @
2007 Dec 21
3
error installing gems
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.
2007 Dec 01
1
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 of Computer Science
Columbia University
Tel: 1-646-387-5998