Displaying 20 results from an estimated 7000 matches similar to: "Can't hear any sound (This time in plain text)"
2008 Feb 05
4
Cannot hear voice through SIP Phone from one side
I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make
calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice.
They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice.
The microphone and speakers are
2004 Oct 05
1
Why I don't hear Call Progress
I'm using sipgate.de as my sip provider. When I'm using xlite as client
on sipgate.de, everything works fine: I call number, hear ringing (real
progress tone form called party, not one generated in xlite) and then
talking with called person.
But, when I'm using Asterisk as sip client on sipgate.de, I don't hear
progress tones: I hear only one (locally generated) ring tone, and
2010 Dec 07
1
[headset/mic] Volume too low + echo in *
Hello,
I'm having the following problem when using a headset on XP
connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus
motherboard:
- Using any sound recorder (Windows', Audacity, XLite), the level is
just too low when speaking at a conversational level, even with the
microphone level pumped all the way up (line displayed totally flat in
Recorder)
2010 Dec 22
4
Asterisk hangs up call after 20s
Hello
I have an Asterisk 1.4 server and two XLite softphones, where
Asterisk and the local XLite phone are located in a LAN behind a NAT
router, and the remote XLite phone is located elsewhere on the Net
behind its own NAT router:
http://img252.imageshack.us/img252/3749/asterisknat.png
I'm having the following issue: When the _local_ XLite calls out the
remote XLite, everything works fine;
2005 Mar 21
4
Can't hear the caller
Hi,
I've got a strange issue, that I haven't found addressed on the wiki.
My asterisk box is behind a firewall which routes udp/tcp requests on 5060 and
8000 to asterisk.
When I make a call from a Zap or SIP extension on the inside of the firewall
to any Zap or SIP extension on the inside of the firewall, everything works
find. I have access to voipjet, and when I place a call
2003 Sep 09
5
Xlite = no sound
What's the secret to getting sound through Xlite? The SIP messages all look
OK to me, but the sound isn't coming through.
It was trying to use GSM, so I searched the archive and tried:
disallow=gsm
allow=ulaw
Now it says that it's using ULAW but I still get no sound in either
direction.
Phil Skuse (MBJEJPIEUI) <phil.skuse@vicorp.com>
2007 Mar 09
0
Fwd: Can't hear any sound
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2006 Mar 07
1
Setting Vaaibles
Helo List,
First I would like to apologize for my bad spelling as
well as that I did not search the wiki first. I only
have email access at the moment.
I am having trouble setting both variables and global
variables thru an extension.
I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4
with an Xlite softphone. I have two xlite phones on
diffent computers. One logs in as xlite1 and the other
as
2006 Oct 19
3
say Asterisk to answer
Hi list,
I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk.
One call the other-one, is it possible to order Asterisk to force answering
the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to
Asterisk which force answer, so Idefisk answer the call without clicking on
"Accept" button.
Greg
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An
2005 Jan 27
2
Soft phone sound quality help
Anyone got any tips on improving sound quality on soft phones running
under Window XP SP2?
I have tried Xlite, SJPhone and Firefly.
They all seem to have significant sound quality problems. We have a
reasonable sized network of several hundred devices connected together
using Layer 2 switches, i.e. pretty dumb switches with no QoS.
I also have a Grandstream connected to the same switching gear.
2006 Mar 15
2
Help with Gizmo from outside firewall
I've beaten myself bloody dealing with this one... No luck so far. In
summary, incoming calls from Gizmo establish, but neither get nor send
sound. Outbound calls to Gizmo work fine (well a bit choppy but work)
My thought is that the SIP connection is being made fine, but the RTP
is getting stopped / blocked / misdone somewhere.
Here is the thing:
Asterisk 2.5 on Linux
(No hardware
2005 Feb 03
2
Odd behaviour between Grandstream and Xlite
Hi,
I've got an Asterisk box with grandstream and xlite clients on it.
No here's the thing:
- I grey out all the codecs on the Xlite except for GSM
- I call the Grandstream from the Xlite, the Xlite uses the GSM codec
and the Grandstream uses ulaw, with Asterisk doing the conversion,
everything fine
- I call the Xlite from the Grandstrea, the Xlite ends up using the
ulaw codec as
2004 Jan 02
4
one way choppy sound problem !
Hi all,
I have my asterisk setup as following:
IP 2 x E1
x-lite <-------> Asterisk -------> PSTN
When I place a call from x-lite to PSTN, the quality of the sound in the
direction x-lite -> PSTN is very bad. That is, the voice of the x-lite user,
heard by the PSTN user is choppy and makes communication not very pleasant.
The sound is choppy as if bits of data
2006 Dec 28
1
one way rtp stream (Sent alwax to 127.0.0.1)
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI> rtp debug
RTP Debugging Enabled
-- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack
-- Called snom
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 answered
2006 Mar 16
1
Newbie needs audio help
My first Asterisk install: Debian sarge with the 2.6 kernel, and two
X-Lite soft-phones. I followed the online how-to documents and was
calling between the two soft-phones and calling the demo system with
no problems and had full audio. I then went on to configure the
TDM400P's two FXS modules. I got into that a ways and was having some
success, but no dial-tone when I was off the
2011 Feb 24
1
Using a Virtual IP Line
Hello!
I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured.
I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.
2011 Mar 17
1
[1.6.2.5] Asterisk can't find MOH file
Hello
I thought I had things set OK to have Asterisk play FR files for
prompts and MOH, but for some reason, it still can't find them:
============ ll /var/lib/asterisk/sounds/
drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/
drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/
Note: fr/ contains core + extra + moh as downloaded from here:
2005 Mar 27
1
Asterisk and XLite on same machine (OSX)?
Dear all,
I have tried to run an asterisk instance together with XLite on a single
machine (a PowerBook).
The intent is to take advantage of IAX connections to easily cross NATs
while traveling.
While the IAX setup proved 'easy', just having to fiddle a little with
working configs at both sides, I did not succeed so far in getting XLite
to connect to the local Asterisk server, AND be
2006 Feb 26
2
Skype vs. an Xlite registered to Asterisk
I have a bunch of road warriors who I've set up with Xlite clients.
Unfortunately
the sound quality has been intermittent at best. Sometimes it's great other
times completely unusable. When it's bad one usually hears harsh static
when the other party speaks or their voice gets "clipped" to static if they
speak too loudly.
Many of these users have migrated to Skype ? much
2010 Dec 06
1
[3102] How to rewrite CID name + number?
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays whatever name I
used in the 3102 and the extension the 3102 uses to register with
Asterisk.
How can I tell