Displaying 20 results from an estimated 2000 matches similar to: "trying directrtpsetup"
2008 Nov 10
3
directrtpsetup without reinvite
Hi,
I want to be able to bridge two sip channels using direct RTP
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:
[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes
[test2]
type=friend
host=dynamic
username=test2
dtmfmode=info
context=test_rtp
2010 Sep 27
1
propagate sip reinvites with directrtpsetup=yes
is there a trick to get asterisk (1.6.2.13) to propagate
codec-changing sip reinvites when directrtpsetup=yes?
i'm trying to route calls to a gateway without keeping asterisk in the
rtp stream.
the gateway is first routing the call to a media server. when
connecting the call to the downstream carrier a different codec is
selected.
the reinvite makes it to asterisk but asterisk isn't
2008 Jul 28
2
Callcentric Issues
Hey,
I have a few dids with callcentric. They seem to work fine most of the
time but at some points I get "handle_request_invite: Failed
to authenticate user <sip:PSTNnumber"
This happens intermittently.
The way I understand it the insecure=port,invite should tell asterisk
not to authenticate users coming from that host. But its not working for
some reason.
This is my sip.conf
2008 Aug 09
1
how to know what codec is being used
Hi,
how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all.
i unset all codecs on x-lite except ilbc.
i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no transcoding happening on the call? call is going thru, rtp is also going thru. what i would like to know is does
2005 Mar 16
1
Re: [Serusers] ser+asterisk - security
Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN).
You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified.
----- Original
2003 Oct 29
1
constrOptim doesn´t send arguments to optim!(?)
Hi,
I think that there something wrong with the 'constrOptim' max/minimization
function because she doesn?t send extra arguments to 'optim' call.
Fact: When I use optim in a f(x,theta)-like function, everything goes ok.
But using constrOptim with the same function leads to error...
Proof: Make a small change in the 'Rosenbrock Banana function' (taken from
the Examples
2004 Jan 23
3
Problem installing Asterisk with Mandrake 9.1
Hi All,
I am trying to get Asterisk up and running on my new Mandrake 9.1 install.
I've installed Linux in the "standard" mandrake security mode, and "su" to do
my attempts at install.
I managed to obtain the source from CVS, and have been able to compile Zaptel.
I then ran insmod zaptel, and also make config.
I think I have compiled and loaded Zaptel successfully as
2007 Aug 06
1
CDR/MySQL basic config
Hi,
I'm trying to add mysql CDR onto a vanilla Asterisk 1.2 install. The
add-ons pack has been installed for a while, so now I'm trying to add
the Mysql config.
I've created a mysql database, added the grants for a user acces, and
can run a mysql -u asteriskcdruser -p and can connect to the database.
I've been using this as a guide:
2010 Aug 24
2
Attempted SIP connection by foreign host. Help!
Say,
I just picked this up on my messages!
There are a whole host of these requests!
Anyone know whow there people are? Is there a way to report them?
Any suggestions as to how to block them?
[Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '"912" <sip:1 at 41.1.1.1>' failed for '184.106.217.112' - Wrong password
[Aug 23 10:34:16] NOTICE[1010]
2005 Feb 02
1
Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients
Hi,
I've spotted weird crash of Asterisk cvs Stable. I have defined queue in
queues.conf :
[prodaja]
music = default
announce = queue-markq
strategy = ringall
context = from-pstn
timeout = 15
retry = 5
maxlen = 0
announce-holdtime = no
announce-frequency = 30
announce-holdtime = yes
monitor-format = gsm|wav|wav49
monitor-join = yes
eventwhencalled = yes
member => Agent/1000
2008 Dec 01
2
Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
Please help.
Asterisk 1: Sip.conf
[VoipDirect777821]
type=friend
host=dfvvd.dyndns.org
username=VoipDirect777821
secret=xxxxxxxxxxxx
accountcode=5260477782
amaflags=billing
context=Incoming
disallow=all
allow=g729
;allow=alaw
;allow=ulaw
trunk=no
qualify=yes
qualifysmoothing=yes
nat=no
canreinvite=yes
dtmfmode=rfc2833
;directrtpsetup=no
t38pt_udptl = yes
Asterisk 2 sip.conf
GNU nano 1.3.12
2008 Oct 03
1
DTMF issues...
I am having a big problem with DTMF. I have a customer using an
Asterisk 1.4.20.1 system with ZTDUMMY as the timing source. The problem
is that when they dial into a conference bridge or IVR where they have
to enter a code they always get an error. Either some numbers are
duplicated or missing.
They use Teliax for calls to the USA and Protel in Mexico. Both
carriers have the same problem so
2015 May 11
2
"Retransmission Timeout" results in dropped calls after 32 seconds
Andrew Martin wrote:
> ----- Original Message -----
>> From: "Joshua Colp"<jcolp at digium.com>
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users at lists.digium.com>
>> Sent: Monday, May 11, 2015 12:32:06 PM
>> Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32
2004 Feb 03
3
sementation fault with mpg123
I'm still getting a sementation fault with mpg123. I have tried different
parameters creating mp3s the last from cd audio ...
lame -m s --resample 8000 -q 0 -a --cbr -b 32
and several versions of mpg123. I have always created 8000 hz outputs. I've
got other * boxes that don't use moh that have been up for months. This one
crashes every couple of days - the verbose output leading to a
2009 Aug 27
1
Bad Gateway
Hey guys,
I've been having a very odd problem that happens intermittently. I've
had this happen with only a couple of providers and somewhat rarely but
its to the point now that we need to fix it to be able to do business.
The scenario is as follows: We have a DID provider that routes calls to
our asterisk boxes and we have an outbound provider to whom we send the
calls of the person
2012 Oct 31
2
Asterisk and OpenLDAP
Hello guys,
i would like to implement authentication for my sip extension with an
openldap server.
Following this guide
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
i see a template named [sip] to map the information of sip peers into ldap.
But i'm not interested to create a template, i would only authenticate
sip extensions using username
2004 Aug 06
6
icecast to rebound streaming radio
My name is Stuardo, from Guatemala and i have this problem: We want to stream
a local radio, but the bandwidth here in Guatemala is no so cheam ass in USA,
so we have a server in USA.
What we want to do is this: to stream the local radio from Guatemala to USA,
this means that the local machine will only have 1 client. Then we want that
the server in USA "rebounds" what it recives
2014 Jul 02
1
Webrtc Not acceptable here
Hi,
I am getting
*Can't provide secure audio requested in SDP offer*
with sipml5 client hosted on my local system
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=sameer ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF
2005 May 04
6
Segmenting wireless traffic
I''m considering adding a seperate subnet to my loc network making loc1
(192.168.1.0/24) and loc2 (192.168.0.1/24) with the goal of seperating
my wireless traffic from my wired LAN traffic.
Has anyone had success doing this, or is it still possible to sniff the
traffic of a seperate zone on the same interface with tools like ettercap?
2019 Jun 14
2
Early Media Issue
Hi all
I've got an issue where when I call a number that just plays early media
back to me.
Instead of hearing the full sequence of tones I hear a short ringing then
part of the sequence. What seems odd is that I can see
the telephone-event/8000 being passed up the chain but when it gets to
Asterisk, it is never sent back to the phone. Instead I just see the usual
RTP flows.
I've been