Displaying 20 results from an estimated 6000 matches similar to: "G711u sound quality decrease with upgrade from 1.0.7 to CVS-HEAD?"
2004 Dec 04
1
Codec translator problem (g723.1,ilbc => alaw)
Hi, I cannot get SIP channel working with folowing codec configuration:
[sip]
disallow=all
allow=g723.1 ;I need this codec between sip phones (BT100)
allow=ilbc ;Use this codec to others
Calling between BT100 SIP phones is OK - asterisk makes native bridge
(with g723.1) between them.
When I'm calling from SIP to other channel (iax,zap,...), asterisk is
not able to chose right codec
2004 Jul 06
3
H323 channel
Hello everybody,
my * box is connected to gnugk with H323 channel. If I call from an H323
EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio
start but noisy (scratch) , then became ok for callee (SIP EP) but still
scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323
EP and it's ok. And from now, it's also ok when H323 EP call SIP one's!
No
2007 Jan 11
1
Has been working for 9 Months - Very Very Strange I cannot dial specific extensions from my dialplan - NOT A CONTEXT PROBLEM!!
Hi all,
I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly
i cannot dial extensions 4XXX from SIP Phones.
Now comes the wired stuff... I can dial this extensions from IAX phones as
well as from Analogue extensions connected to our legacy pbx, that is
installed on front of asterisk.
So :
Zapata Calls to SIP extensions 4XXX - OK
IAX to SIP 4XXX-OK
SIP to SIP 4XXX -
2003 Jun 17
4
soft phones -- voice quality tuning
I've got the XTEN Lite soft phone mostly working with * but it's
dropping out like a very bad cell phone call.
The GSM codec is worst (unusable), G711u and G711a are best but
not good enough to use.
I don't think it's a lack of bandwidth.
What tuning options or approaches should I be investigating to
make this work.
Also, what's the best soft phone(s) for Windows XP?
2004 Dec 07
3
:: Migrating to 1.0.3 => Attention. ::
Hello list ,
I?d like to announce possible problems with migrating any version prior to
1.0.2 to 1.0.3.
Pay attention :
1. Codecs
Codecs names/description have been changed .
For example :
versions <= 1.0.2
voip*CLI> show codecs
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
1 (1 << 0)
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2005 Feb 16
3
HELP!!!!!!!!
Hi,
I have installed two X-Lite phones and they're able to login successfully.
The two phones plus the Asterisk system are all on the same LAN with private
addresses assigned to each of them. When a call is initiated and is picked
up on the other end, there is completely no sound at all (as in the line
goes dead). The codecs set in the softphones are g711u, g711a, GSM, iLBC and
SPX.
2005 Feb 09
4
IAX Voice Quality Issues
I am running * 1.0.5 and have been having lots of problems with
outgoing calls and their sound quality. I am using ULAW for the codec
and sixtel for termination. Basically the problem is that portions of
the call seem to be lost and replaced with silence. Sometimes I can't
hear the person talking othertimes they can't hear me. This situation
comes and goes throughout the call. Bandwidth
2007 Nov 15
1
Help on strange problem...
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hey all,
I'm having problems with calls dropping after 15 - 20 seconds from a
particular provider. The are using a NexTone gateway. Here are the details:
Successful call:
INVITE cseq 1 From NexTone
100 Trying cseq 1 From Asterisk
100 Trying cseq 1 From Asterisk
200 OK (G711U) cseq 1 From Asterisk
ACK cseq 1 From NexTone
INVITE (G711U)
2004 Jan 06
1
Fw: Pls confirm
----- Original Message -----
From: "Jess Magnaye" <jess@arretni.com>
To: <wipe_out@users.sourceforge.net>
Sent: Tuesday, January 06, 2004 3:19 PM
Subject: Re: [Asterisk-Users] Pls confirm
> Is the format "allow=g723.1" in sip.conf valid?
>
> somehow i cannot get it working to do g723 passthru. also, i've read that
> doing g723 will disable
2015 Nov 20
2
SIP calls dropping at 15 minutes
I have a problem where SIP calls from some providers are dropping at 15
minutes.
The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS
sends calls to an Asterisk server.
Below,
'Client' is the IP address of the client's host (running
FPBX-2.8.1(1.8.20.0)
'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls
'Asterisk' is the IP
2005 Feb 01
0
chan_capi and G711u
Hello all,
I've got an AVM Fritz!Card PCI that I'm using with Asterisk under
chan_capi (0.3.5)
Phones on our internal network all use g711u. I'm aware the chan_capi
uses g711a by default.
To reduce the need for transcoding, I decided to make everything use
the same codec.
First I changed all the phones (Cisco models 7905G, 7940G and ATA 186
all running SIP) to g711a, but it seemed
2003 Oct 23
1
How to write sound file with G723.1 codec or G729 codec
Hello, all
How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf
[writesound]
exten => s,1, Answer
exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729)
I'am using oh323 channel driver, in oh323.conf
2007 Jun 13
3
WAV file best sound quality
Hi,I have been using wav files with sample rate of 8khz and 8 bits and I find the sound quality really poor.What is the best sound quality I can achieve on Asterisk?Responses would be appreciated.Rgds,Akpome
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2003 Aug 13
1
FWD SIP phone format=2, FWD call format=4, why?
Hi!
I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the
IP phone. I configured the X-Lite and SIPPS to use GSM codec. Whe I
call FWD, I get this info on the channels when the call has not been
stablished yet:
sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter
Format
192.246.69.223 613 1770bf3430d 00102/00000
2009 Jul 09
1
Connecting two Asterisk together via SIP + DISA
Hi all,
I need to test the following scenario:
+-----------+ +-----------+
| asterisk 1| | asterisk 2|
+-----------+ +-----------+
| |
| |
_______|__________________|___________
| |
| |
| |
+-------+ +-------+
| ATA 1 |
2003 Sep 19
1
codec probs wit g723.1
Hi all,
i don't know how often someone ask for this, but i ask agian:
Is it possible to use G723.1 with * or not ?
I tried to use G723.1 from * over OH323 to a gatekeeper from my provider.
The situation is following:
Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER
The provider supports G723.1.
Can someone help me ?
Regards,
Thomas.
2013 Jun 02
1
Issue in transcoding
I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm
gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have
g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1
and call leg from gsm gateway is using codec gsm. I am having one way audio
and getting below mentioned warning. Asterisk version is 1.8.11.0
[Jun 2 17:08:28] WARNING[21652]:
2011 Feb 10
2
Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.
I have an asterisk
2004 Jun 30
3
Bugfix for CVS-HEAD-06/26/04-21:56:45
Hiya,
I sent this bugfix to the asterisk-dev mailing list, and modified it as I
noticed side effects, but now it appears to be finished. Nobody seemed to
notice it there, so I thought I'd post here, as it seems to be something
that will be needed as people update to the latest CVS version. So...read
on :)
Ted
programmer_ted@hotmail.com
P.S. Read to the very end. The original bugfix