Displaying 20 results from an estimated 11000 matches similar to: "Asterisk Audio Problem"
2004 Oct 06
2
no audio from asterisk
I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57.
I have tested both KPhone and IaxComm for linux but receiving no audio
from asterisk.
sound is working fine, as I can listen playing files using PLAY or
APLAY.
KPhone is configured with DTMFmode=inband and codec is ulaw
and IaxComm is configured with ilbc
if somebody can sort out this
Thank you
regards,
--
Atif
2003 Apr 16
4
iLBC
i tried asterisk ilbc codec against kphone. when the call got
connected, i started to immediately get these kind of message to the
console:
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)?
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of
2005 Jul 21
0
kphone & Asterisk CVS HEAD: no audio
Dear Asterisk experts,
I've just downloaded Asterisk CVS version (since I'd like to manage
its configuration from RealTime).
Next, I have kphone on the same Linux host, and VmWare virtual
machine with Windows and X-Lite IP phone inside.
I successfully tested the demo's with X-Lite, but failed to hear
something with kphone (v4.1.1). There were NO problem with this
kphone and stable
2004 Sep 30
2
OT: Kphone installation problem
Hello,
I know that my Kphone question may be a bit off topic, but I have been
busy with this again and again for about one month now, sent three
mails to kphone@wirlab.net (the contact address mentioned on
http://www.wirlab.net/kphone/index.html), asked for a solution in a
german ip phone forum and tryed many things by myself.
I try to compile KPhone 4.0.3 (tryed CVS Version as well) but
2005 May 17
1
sip show registry empty ?!?!!?
Hi all,
i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones)
and this is what my "sip show users" return:
moloch*CLI> sip show users
Username Secret Accountcode Def.Context ACL NAT
204 moira from-internal No No
203 michele from-internal No
2004 May 25
1
Troubles with Kphone
Hi ,
I'm triying to use kphone 4.02, but when i'm make a call the programs
doesn't respond any command, so i can't hear any sound ..
in sip.conf that's my codec config:
disallow=all
allow=gsm
allow=ulaw
allow=ilbc
and the kphone give the follow :
SipClient: Sending: 06:46:28.116
--------------------------------
ACK
2005 May 18
0
HELP ME!!!! Asterisk don't do calls
Hi all,
as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions:
moloch*CLI> sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN
203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms)
202/202
2005 Feb 17
1
(Kphone) Registration Failed: Forbidden
I just can't get kphone to register with asterisk, i can make calls to
the demos and even get into the mailbox but kphone cannot register.
Here's my story. Can you help me?? Please
I have installed asterisk on debian using apt-get install asterisk.
I have configured an extension in extensions.conf as follows
exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt)
exten =>
2004 May 25
1
Troubles with Kphone]
-------- Original Message --------
Subject: Re: [Asterisk-Users] Troubles with Kphone
Date: Tue, 25 May 2004 15:44:15 +0530
From: Murali Krishnan <murali@bksys.co.in>
Reply-To: ismk@myrealbox.com
Organization: bk SYSTEMS (P) LTD.,
To: asterisk-users@lists.digium.com
References: <200405250652.46370.klky3@fibertel.com.ar>
enano wrote:
>Hi ,
>
>
>
>I'm triying to use
2003 Jun 09
0
iLBC, Speex and X-Lite
I've been trying out the newest X-Lite (Build 1012) with iLBC and speex
codecs.
If I enable only iLBC _or_ SPX on X-Lite and call the echo-test on my
asterisk server, the call connects, but I get no sound.
If I enable only iLBC _and_ SPX, X-Lite indicated that it has connected
with iLBC, and I hear a weird squawking.
My sip.conf contains:
allow=iLBC
allow=SPEEX
allow=gsm
I've heard
2003 Sep 17
1
core dump back trace of chan_oh323
hi michael,
here are the core dumps.
only kphone works when 0.5.5 and * cvs.
audiocodes and msn messenger all cause seg faults
when calling ccm thru * (or vice-versa)
~kelvin
[chan_oh323.so] => (OpenH323 Channel Driver)
== Parsing '/etc/asterisk/rtp.conf': Found
== Parsing '/etc/asterisk/oh323.conf': Found
0:00.004 OpenH323 Wrapper OpenH323 Wrapper
2004 Jul 13
1
segmentation fault on asterisk startup
Hi,
I write to this list, because I didn't find anything on the net.
I installed asterisk using bristuff-0.0.2 without any errors, but when I
start asterisk with "asterisk -vvvc" I get following error:
[codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator)
== Registered translator 'ilbctolin' from format ILBC to SLINR, cost 127
Segmentation fault
Removing
2007 Aug 29
2
sip authorization problem
Hi,
I am trying to setup a simple home voip service w/ *
I have compiled and installed the svn source
as a first step I am trying to configure SIP for inside my network.
I have a handful of softphones and a few hardphones that I want to all be
able to call each other
I have configured users.conf with a single softphone(kphone) and have tried
calling itself (ext 6000) and the demo
from the
2005 Mar 23
0
Local sip client stuttered audio
I have asterisk running on my personal computer and am using Kphone to
connect to it. My provider is broadvoice which is Ulaw and I had kphone
connected as GSM. The lag was terrible coming from
Pots-->--Broadvoice-->Kphone. About 2.5 seconds! Going the other
direction seemed fine. I did a:
show translation recalc 200 and see that the translation time should be
about 2 ms. When I do the
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on
2005 Jun 09
0
Conversations cuts: "didn't get a frame from Channel: SIP/..."
Hello list,
I've been looking around to solve this problem both in the IRC, the wiki
and Google without any luck.
I have a box running * 1.07 (with zaptel 1.07 and libpri 1.07) and I'm
having massive problems with random and increasing conversations cuts
both in Zap and SIP channels.
When one starts a call or attend any, sometimes it cuts-off at 20-25
seconds without any apparent
2004 Jul 15
1
Fedora Core 2 softphone
Hello all,
I am in the process of converting our company over to * to replace our
ancient executone system. As part of the testing process my boss wants
us to all run softphones on our desktops until he gets the phones
ordered. Quite a few of us run fedora core 2, and I haven't had any luck
getting a soft phone to work. Kphone works the best out of all I have
tried but I get no sound out of
2004 Jul 23
0
SIP - Cancel request fails with "481 no such call"
Hi,
I am using SIP extensions connected to the PSTN with the CAPI Channel
driver.
All works fine except that one of the sip phones keeps ringing when the
caller
hangs up before extension is answered. The phones are grandstream 100,
though
we get the same behaviour using other phones (X-lite, Kphone).
It behaves the same regardless of whether the incoming call is from a SIP
extension or an
2004 Jul 09
0
GSM to iLBC one way audio :-(
Hi,
I'm using IAXphone for remote users which limits me to the GSM codec.
Internally I limit the SIP phones to iLBC codec (GS 101 1.0.5.0)
I also use voiptalk.org for external PSTN access again using the iLBC codec.
The problem I have is that when the IAXphone dials an internal phone or PSTN number either the line hangs up immediately or there is only one way audio from IAXphone.
It works
2003 Aug 04
1
SIP clients not sending audio
Hi, I've got two SIP clients, one is X-Lite on NT, the other is
KPhone on Linux and when I try either the echo test or voicemail
demos, they fail to send any audio. They are both set up as type
of "friend" in sip.conf so that they can send and receive calls.
Using an IAX client like Gnophone, I have no problems.
The troubling thing is that I'm almost certain that this was