Displaying 20 results from an estimated 3000 matches similar to: "Outgoing call with bad/choppy sound"
2008 Apr 25
1
choopy audio when both side talk at the same time
Hi
I have a server with the last version of asterisk branches, zaptel
branches, 2 Digium Card with TDM800P
16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10
Grandstream GXP2000.
zapata.conf
echocancel=64
rxgain=0
txgain=0
when i place a call o receive a call, I finish a sentence i hear a
ssssssss, AND when the both side talks at
the same time i have choppy audio.
Any
2002 Aug 29
0
PATCH: Fix IRIX 6 testsuite failures
Having built rsync 2.5.5 on IRIX 6.2 with gcc 3.1, I ran into two failures
when running the testsuite with make check: both the chgrp and hardlinks
tests fail:
The failure in the chgrp test occurs here:
+ rsync -rtgvvv /amnt/callisto/volumes/obj-irix5/local/obj.irix5/rsync-2.5.5/testtmp.chgrp/from/ /amnt/callisto/volumes/obj-irix5/local/obj.irix5/rsync-2.5.5/testtmp.chgrp/to/
rsync: opendir
2002 Jun 15
0
[PATCH] make test patch.
Yo,
Currently if you build rsync and you don't have rsync installed, make
test will fail because some tests reference the rsync binary directly
(instead of using the $RSYNC variable).
I cannot find anything which refers to this problem in the mail archives
or CVS logs.
Below is a patch which fixes this problem.
(please CC: me, I'm not subscribed. Thanks.)
-- Geoff.
--------------
2002 Oct 08
1
Some tests fail if rsync is not on path (with patch)
While installing rsync on a new Sun Netra running Solaris 2.8, two tests
(chgrp and hardlinks) failed.
I found that these tests execute rsync while other successfull tests
exectute $RSYNC. It is fortunate that my shell path was quite restricted
and that no earlier version of rsync was installed on my path. The system
would have run the chgrp and hardlinks tests with an earlier rsync if it had
2004 Dec 23
1
Can't Make Outgoing Call
Hi,
I can't get dial-out working. I'm trying to call 523936. Is there
something wrong with my setup here? Could someone please give me a few
pointers?
Regards,
Norman Zhang
[fwd-out]
exten => _8.,1,SetCallerID(${FWDUSERID})
exten => _8.,2,SetCIDName(${FWDUSERNAME})
exten => _8.,3,Dial(SIP/${EXTEN}@fwd,70)
exten => _8.,4,Macro(fastbusy)
exten => _8.,5,Hangup
*CLI>
2004 Oct 07
1
Confused about NAT and Authentication with FWD
I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT.
The problem:
I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem.
Now if in the extension.conf file I have,
exten =>
2007 Jul 20
2
Announcing Digium/Asterisk World's Conference Program
Is this replacing Astricon this year?
If so it looks like a pretty poor showing in comparison to Astricon
Dallas last year.
Cheers,
Dean
________________________________
From: Carl Ford [mailto:carlf at vonmag.com]
Sent: Wednesday, 18 July 2007 9:09 AM
To: Dean Collins
Subject: Announcing Digium/Asterisk World's Conference Program
2003 Aug 12
0
RE: Asterisk-Users digest, Vol 1 #1033 - 7 msgs
Same thing. It will make sense to try
Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082
but in that case Asterisk sends
REGISTER sip:fwdnat.pulver.com SIP/2.0
which is not right. It should be sip:fwd.pulver.com but sent thru
fwdnat.pulver.com:5082
BR Borut
-----Original Message-----
Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT
From:
2003 Aug 19
1
Problem with * server and FWD
I have a small HUGE problem with *.
I have installed * but I have 2 problems.
1 - Making call to FWD.
2 - Receiving call from FWD
More info of the problem at the end.
Here is the sip.conf file.
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = sip ;default Default for incoming calls
register =>
2004 Dec 02
0
Connection Problem
Hi,
My configuration: Sipura 2000
Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux
I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for the first second the connection seems to be ok and I hear: <thu .. rsda ..> and nothing more, which suppose to mean <Thursday, ...>. The echo line from 613@fwd.pulver.com does work the same. I never
2003 Dec 20
2
More beginner questions
Using DIAX softphone which seems to be working OK can get to VM/echotest etc
in the demo context
Am trying to setup FWD but get the following problems
Can hear it ringing when dialing FWD no 612 for time. Connects but no sound
from remote end.
Does anyone have any suggestions.
Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to
the internet port 5060 being forwarded to
2003 May 31
0
register with outbound proxy from behind nat for freeworlddialup etc.
Hi,
I've posted a simular message little over a week ago so sorry for
reposting. I need to register to freeworld dial up from behind a nat.
Using the xten software sip client works fine but with asterisk I don't
know how to do it. Last time I posted I got different responses. Some
saying I can't register with an outbound proxy from asterisk others said
they have done it. If it is
2004 Jan 17
1
Registering multiple FWD accounts
Can multiple FWD accounts be registered?
I have the following output in my sip.conf file:
register=74928:xxx@fwd.pulver.com/74928
register=75160:xxx@fwd.pulver.com/75160
register=74573:xxx@fwd.pulver.com/74573
[fwd-74928]
type=friend
secret=xxx
username=74928
host=fwd.pulver.com
[fwd-75160]
type=friend
secret=xxx
username=75160
host=fwd.pulver.com
[fwd-74573]
type=friend
secret=xxx
2004 Sep 14
1
Setting up Asterisk with fwd
Hey all,
I'm trying to get my Asterisk server up and running on
fwd.pulver.com just to get the hang of it until I get
my FXO card in a couple of days. It seems to connect
but that's about it. If I try to dial into it from
another fwd # it says user is not online.
In sip.conf I have the following added:
register => xxxxxx:xxxxxx@fwd.pulver.com/489125
[fwd.pulver.com]
type=friend
2003 Sep 28
1
Forwarding SIP over IAX problem: No One Available
I'm hoping someone can help me out with this. I am basically just
trying to separate my outbound and inbound calls into separate contexts
instead of having everything in a single context. Any help would be
appreciated. Perhaps I've missed something really obvious....
Here is the network layout:
<remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>
2009 Jan 25
2
Choppy Sound On Bridging From SIP->IAX
I am experiencing choppy sound when I bridge from a SIP peer to an IAX
peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am
experiencing choppy sound from the SIP peer to the IAX peer but not
vice-versa. I know that this is not a bandwidth issue because I don't
have choppy sound (with the same codec) when bridging IAX->IAX peers or
SIP->SIP peers. My timing source is
2007 Aug 06
0
Digium|Asterisk World
Too bad it is August 6th
*P.S. Remember, as a member of the Digium Family we have secured a
special discount of 50% off of the conference fee for you if you
register by July 29, 2007. To take advantage of this limited time offer,
please register here
<https://secure.pulver.com/cgi-bin/von?mode=gpur&conf=dawfal07&type=g&pricode=billm>!*
Digium, Inc wrote:
>
> If you
2003 Nov 28
0
Can't seem to connect/call fwd network Help!
I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various
2007 Sep 06
1
Choppy sound while converting alaw to ulaw
Hi there
I europe alaw is usual. I have a SIP Phone which perferes ulaw.
When my * box has to transcode alaw to ulaw the sound get's one way choppy.
(alaw => ulaw is choppy, ulaw => alaw is fine).
I managed to fix the issue by forcing my SIP phone to use alaw only, but is
this a know issue with asterisk 1.2.13?
-Benoit-
2009 Feb 17
2
Packet Truncated - Choppy Audio
Hi there,
We're having some complaints of choppy audio from our SIP customers.
Asterisk is showing no errors, but I'm getting a lot of these in my syslog:
Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654->8232)
The first number varies, but the last number is always 8232.
I've read that this is a common MTU size, but none of our interfaces
have an MTU of 8232.