Displaying 20 results from an estimated 300 matches similar to: "Getting 'username' of sip peer"
2005 Jun 05
2
Problem in the path for executables...
Hi,
I have a windows executable which I am trying to run in Linux using wine.
When I execute the command :
wine {ABSOLUTE_PATH}/file1.exe,
file1.exe runs many other executables internally, like file2.exe, file3.exe
and file4.exe.
Now when file1.exe is trying to run the other executables, it is not able to
get the path of the executables. I have the "PATH" enironment variable set
2006 May 23
1
Problem with options to "Dial" application
I'm trying to set a dialing rule in my dialplan. As a part of it, from
my point of view, this works wrong
priorityjumping=no
[test_context]
exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag
exten => 1234,2,Playback(digits/2)
exten => 1234,3,Playback(digits/3)
exten => 1234,102,Playback(digits/4)
In this case, if I dial the extension, and
2006 May 30
20
AEL #include
Anyone know if #include works in ael yet?
extensions.ael:
#include "inc/pbx/global.conf"
context test_context {
};
*CLI> ael reload
May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include'
May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged
2010 Jun 26
2
Detecting hook flash in asterisk
Hello,
Is it possible to detect a hook flash in asterisk. I want to be able to
perform some functions an hook flash.
I have the following entry in features.conf which executes a Macro on
detecting key press '**'.
[applicationmap]
test => **,caller,Macro,testflash
Is it possible to do this action on hook flash?
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2010 Jul 01
3
Originate multiple channels
Hello,
Is it possible to use the asterisk manager interface to originate
multiple channels?
like
Action: Originate
Channel: SIP/101&SIP/102
So that both extensions 101 and 102 rings simultaneously.
I am using asterisk manager interface over http.
Thanks
2015 Jun 12
1
Voice mail and caller ID
On Fri, 12 Jun 2015 11:49:05 -0700
John Kiniston <johnkiniston at gmail.com> wrote:
> Try this for CHAN_SIP:
>
> same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
> same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the
> mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we
> have a mailbox defined log into it
Perfect.
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
2015 Jun 12
2
Voice mail and caller ID
I have this in my sip.conf:
exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}")
same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s)
same => n,Hangup
However, my extensions are set up so that they always show the external
number, not the extension:
[foobar2](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx
callerid=Candace <5555551212>
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 =>
2006 Feb 13
1
How to Get SIP Header : To Field ?
Hi,
I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch.
When forwarding a call to Voicemail, here is somehow what the softswitch
sends to Asterisk :
In INVITE : Vm Phone Number ( to route the call )
In To : Person who has been called !
In From : Person who was calling !
Of course, I need to send the call into the "Called User" Mailbox (Thus To
SIP header) !
So
2010 May 21
2
Using unix socket to connect with database
Hello,
I am using asterisk realtime with a postgresql database on the same server.
In res_pgsql.conf I have specified
[general]
dbhost=localhost
dbport=5432
dbname=asteriskdb
dbuser=psql
dbsock=/tmp/.s.PGSQL.5432
Since both asterisk and db are on same server, I would like asterisk
to connect to db using the local unix socket. However asterisk is not
using the local unix socket to connect to
2011 May 02
1
default context overrides context of peer
Hello,
I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17.
I have context=defcontext set in sip.conf. For each peer I have
context=outcontext in the peer definition since I want outgoing calls
from registered SIP peers to go through context 'outcontext'. This
used to work in the older version (1.6.2.7), but after upgrading this
has stopped working. Now outgoing calls are going to
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 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=${SIPCHANINFO(recvip)}
2014 Feb 11
1
file.c:1160 ast_writefile: Unable to open file /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such file or directory
Dear Folks,
[Test_Context]
exten => _911.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _911.,2,Set(CALLERID(num)=xxxxxxx)
exten =>
_911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})
exten => _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})
exten => _911.,5,Set(${CALLERID}=${CALLERID(num)})
exten =>
2009 Jun 03
1
IAX2 Channel Information
I'm trying to isolate the IP address of inbound calls to my switch over
IAX2. Is the proper way to get that information as follows:
${IAXPEER(IP)}
If the caller was inbound via SIP, this works:
${SIPCHANINFO(PEERIP)}
So I'm looking to return the IP address of the caller via IAX2.
Thanks
Lee
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2010 Aug 10
1
DEBUG: Cannot find variable 'XXX' ??
On 1.6.2.11-rc2 I've noticed a bunch of DEBUG statements on startup,
such as:
== Registered custom function 'SIP_HEADER'
[Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot
find variable 'SIPPEER' in tree 'description'
== Registered custom function 'SIPPEER'
[Aug 9 07:01:23] DEBUG[17330]: xmldoc.c:1727 xmldoc_build_field: Cannot
find
2007 Oct 26
1
Can't get sangoma A102D setup on asterisk
I have a new Sangoma A102 and I'm trying to get it running in asterisk. A
look through the dmesg log shows the card is detected and the various
channels created. However, when I start asterisk I get the error below.
Any ideas?
My zapata.conf is below.
Thanks,
MD
== Registered custom function SIPCHANINFO
== Registered custom function CHECKSIPDOMAIN
== Manager registered action
2005 Jun 07
0
Problems in installing wine...
Hi,
We have got wine-20050211-1rh73winehq.i686.rpm package. We ran rpm2cpio
and cpio to get the required binary, library and config files.
We put all these files on a server. Users nfs mount and get the
appropriate binaries. This is done by setting the environment variables :
"PATH", "LD_LIBRARY_PATH" and the "WINEDLLPATH".
when we run wine for the first time,