Displaying 20 results from an estimated 8000 matches similar to: "is it possible to deactivate RTCP?"
2009 Aug 27
6
Measuring voice quality with Asterisk
Hi!
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup with Asterisk - is it possible to get
RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)?
Thanks
Klaus
2009 Mar 06
5
work around the 64 pickupgroups limit
Hi!
What are the typical ways to work around the 64 groups limit?
thanks
klaus
2006 May 11
3
sangoma A102 installation question
Hi!
I've went through the READMEs and could not answer this question:
During installation, the Setup program asks:
Would you like update/upgrade wanpipe drivers? (y/n)
For a pure Asterisk TDM installation - is it required to patch the
kernel or is this only when using the sangoma cards as WAN router?
regards
klaus
2009 Jan 14
8
evaluate SIP response codes in dialplan
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards requests to various PSTN
gateways with SIP. If the Dial() attempt is not successful I want to
differ at least these 3 options:
- called destination is busy (486): e.g. activate auto-redial
- called destination does not exist, unassigned number (404)
- gateway is broken,
2010 Sep 22
5
http://www.asterisk.org/downloads naming schema
Hi!
Since some time the download of the newest Asterisk does not contains
the version number anymore, but is just called "asterisk-1.4-current.tar.gz"
This gives me a tarball where I do not know the version without looking
into the tarball.
Thus, IMO it would be very useful to switch back to old schema war the
download contained the version number.
Thanks
Klaus
2009 Nov 10
2
looking for an Asterisk supervision (status viewer) tool
Hi!
I am looking for a tool (application or webinterface) which shows me the
current status of an Asterisk server, e.g.:
- Status of the SIP peers (registered/offline)
- current incoming and outgoing calls
- start-time, numbers, some history
- history (calls stopped in the last 15 minutes, who hang up?)
- should be possible to link those calls to the relevant SIP peers
-
2009 Jun 08
3
T.38 pass-through 488 handling problem
Hi!
I have the following problem with Asterisk 1.4.23:
ATA w/ T.38 Asterisk ATA w/o T.38
--------INVITE-------->
--------INVITE-------->
<-------200OK----------
<-------200OK----------
--------ACK----------->
--------ACK----------->
--------INVITE
2009 Jan 08
3
AEL and };
Hi!
All the AEL examples have a semicolon after the closing curly bracket, e.g:
context test {
1 => Hangup();
};
but without ; it works fine too, e.g:
context test {
1 => Hangup();
}
So - what is the reason for the ; after the closing curly bracket?
thanks
klaus
2009 Feb 25
3
Asterisk with Internet connectivity
Hi!
I have a setup with Asterisk in front of a PBX connected with ISDN to
the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing
ENUM for outgoing calls and allows incoming calls per SIP.
Recently the IP connectivity for this location was down the whole
telephony was down too - not even incoming calls did work. This is
really strange as incoming calls from PSTN are routed
2008 Dec 23
2
why does users.conf generate SIP peer and SIP user?
Hi!
I wonder why users.conf generates a SIP user and a SIP peer? Why is it
not possible to set type=... in users.conf? (Asterisk 1.4.22)
thanks
klaus
2009 Feb 24
7
multiple asterisks in a server
Hi all,
Is it possible to install more than 1 asterisk in a single server?
If yes, what do I need to set and take care?
Rgds,
ango
2010 Feb 08
2
conferencing without DAHDI
Hi!
IIRC there was an announcement some time ago that it is possible now to
make conferences without the need for DAHDI anymore - but I can not
remember the name of this feature anymore, and google didn't solved my
problem.
Thus, any references to this new system are appreciated.
thanks
klaus
2008 Nov 13
5
database queries from extensions.conf
Hi!
What is the preferred way to make database lookups from within the dialplan?
I only know the MYSQL function from asterisk-addons. Are the other
methods too? (e.g. for postgresql, unixodbc)
thanks
klaus
2009 Aug 04
4
CDR Problem - No CDRs when call is not bridged
Hi!
I just found out that Asterisk (1.4) does not write CDRs if the incoming
call was not forwarded but handled internally without answering the call.
E.g.:
[from_pstn]
exten => 997,1,Answer()
exten => 997,2,Playback(tt-weasels)
exten => 997,3,Hangup()
exten => 999,1,Playback(tt-weasels|noanswer)
exten => 999,4,Hangup()
For incoming calls to 997 a CDR will be written, but not
2009 Jan 08
4
AEL question: testing channel variables
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=YES
Of course I could use the following code, but this bloats up the code:
if (${EXISTS(${FOOBAR})}) {
2008 Nov 28
1
RTCP too short
Dear Sir,
I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk
-rvvvvv I can see a lot of messages about RTCP too short...
-- Remote UNIX connection disconnected
[Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too
short
[Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too
short
[Nov 28 13:33:00] WARNING[19803]: rtp.c:891
2017 Sep 19
0
AST-2017-008: RTP/RTCP information leak
Asterisk Project Security Advisory - AST-2017-008
Product Asterisk
Summary RTP/RTCP information leak
Nature of Advisory Unauthorized data disclosure
Susceptibility Remote Unauthenticated Sessions
Severity Critical
2010 Jan 28
2
rtp.c:883 ast_rtcp_read: RTCP Read too short
Hi:
I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?:
?
-- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack
??? --
2003 Jul 04
1
How to make * send RTCP reports
Hi,
I am plying with * for 10 days now. I am testing with a couple of vocaltec
h.323 gateways (FXO and PRI) cisco ata-186 (configured for SIP) and MSN
messenger (SIP). They all seem to interoperate. However I have a problem
when * is sending calls to the vocaltec gateways. Vocaltec gateways are
monitoring the RTCP reports send from the remote gateway (in this case *)
and if they don't get a
2003 Nov 18
1
Will Asterisk be supporting RTCP XR in the future?
This article below came up on the newwire. The RTCP XR RFC was published.
Will Asterisk be supporting this function in a future release? Does anyone
know if any phone vendors are going to be supporting it?
Thanks
Lee Goodman
Our Technology Update this week is about one of those
mechanisms. Known as RTP Control Protocol Reporting Extensions
(RTCP XR), the technology defines a standard way to