Displaying 20 results from an estimated 30000 matches similar to: "G723 error"
2005 Sep 02
1
G711u sound quality decrease with upgrade from 1.0.7 to CVS-HEAD?
Hi,
I was running asterisk 1.0.7 but we've upgraded now to CVS-HEAD.
I've noticed this.. and several people have commented that audio
quality seems to have gone down hill. Just going
phone-->asterisk-->PRI. I've not changed the configuration files
during the upgrade.
sip.conf is:
allow=ulaw
allow=ilbc
allow=g726
allow=g729
allow=g723.1
And all the phones had been using
2006 Apr 19
1
Codec problem from SIP to H323
Hello.
I have a codec problem to send calls from a SIP device to a H323 gateway.
First I'll explain the scenario:
- Asterisk 1.2.1
- The SIP phone can use any codec I want.
- The H323 gateway can only use g729 (cause it's not under my
administration)
- SIP phone has g729 configured, so my asterisk doesn't need to "transcode"
(I don't have licences for g729)
- sip.conf
2006 Nov 20
1
g729 registered
Hi guys,
I've registered some g729 licenses, during register process everything
worked fine.
astk2*CLI> show g729
0/0 encoders/decoders of 20 licensed channels are currently in use
But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk
using g729 but I got this answer from asterisk:
Got SIP response 488 "Not Acceptable Here" back from
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
2004 Apr 13
1
SIP->h323 problem DTMF
I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711).
But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message:
-- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack
--
2008 Jan 23
2
Replacement for Allison
Hi,
Does anyone know what I need to do to get these:
http://www.enicomms.com/cutglassivr/
Sounds files to work? I've tried loading them, but they are completely
silent (format mis-match maybe?). Specifically, when I try to enter
voicemail, nothing plays... though it clearly tries.
I'm looking for replacement sound files for the default Allison, as I feel
she is kind of breathy. I have
2005 Jun 06
1
Issue with SIP inter-op
Hi All,
I'm trying to connect to a SIP carrier who never connected with Asterisk.
I managed to connect with a sipura phone or a grandstream, no problem.
When I configure asterisk, I'm able to send out calls to the carrier no
problems,
however, receiving calls doesn't work, and I keep getting the following
messages:
<-- SIP read from 69.xx.xx.xx:5060:
INVITE
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
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)
2010 Jan 12
2
SIP Security
Hey guys,
I've been running asterisk on my server for some time now (currently
running Asterisk 1.6.2.0). I am having security issues with my SIP
accounts. Unauthorized people have been able to access the server (bots)
and they have been able to make calls (in today's case to Cuba).
Here's a copy (slightly modified) of my sip.conf:
[general]
context=default ; Default
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
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 -
2005 Jul 07
1
Calls with oh323 with no sound
Hi,
I've oh323 chan installed and working to make calls from SIP to H323
devices. The problem is can no hear sound with the H323 device. I think
this is some related with codecs o nat, because the H323 have one public
IP from a different subnet from the asterisk box.
If I use netmeeting in gateway mode, the call can be completed and I can
talk with a SIP device, but in gateway mode I can not
2004 Dec 06
1
DTMF via PSTN to * to IAX to * challanges.
Ok I have an * server finally setup and acepting calls from freshtel and
I am VERY impressed at how well the Freshtel.net service works but thats
another subject :)
I have it all setup so that I can Dial my DID number on freshtel and
that gets set to my * via IAX.
At the moment I have the demo configured so that I can test it all and
make sure it is all working.
The problem is that I
2006 Mar 17
3
SIP Realtime Users
Trying to get SIP realtime working here...
I'm connected to the database...
*CLI> realtime mysql status
Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds.
I can get information for the extension in question...
*CLI> realtime load sipusers name 2944093
Column Name Column Value
2003 Oct 23
1
Problems with OH323/codecs
On oh323.conf I have:
codec=G711U
frames=20
But while connecting it gives me in log:
? 1:18.636 ? ? ? ? ?H225 Caller:8111de8 H245 ? ?Capability merge result:
? Table:
? ? G.723.1(5.3k){hw} <1>
? Set:
? ? 0:
? ? ? 0:
? ? ? ? G.723.1(5.3k){hw} <1>
Which I don't have, so the connection is dropped. Any known solutions? (remote
side has g711 u-Law)
--
Witold Kr?cicki (adasi) adasi
2005 Feb 16
1
Passthrough and reInvite
It is not clear how exactly g729 pass-through can be enabled. I
have a SIP call off a gateway come into an Asterisk menu, and then I
send the SIP call to another SIP gateway using Dial(). Even though
codec preferences have g729 listed first, it never gets used.
Both gateways have separate peer entries in sip.conf, and both have
canreinvite=yes set. Can Asterisk change the media type during
2006 Nov 19
1
G723 pass-through and codec negotiation
All,
Our users have a softphone client that supports the G723 Codec which we
want to use for bandwidth reasons, however we do not wish (or have the
funds) to license the codec on our Asterisk servers. We have G723
pass-through working between the clients just fine, however calls fail
when terminating with Asterisk itself (i.e. Voicemail) or out to the
PSTN due to transcoding issues.
If it
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
--
2009 May 19
1
Alternative to Adobe Audition 3 for G723 > G711 uLaw ? (old Cool Edit Pro)
Can anyone recommend a codec pack with G723 for use under Vista? I have
G723 prompts (about 70 prompts totaling 1MB) needing to be converted to
G711 uLaw.
I tried Audacity but it doesn't have G723 codecs. I tired some google
found adware free tools and websites with no success in converting G723.
It does appear the old Cool Edit (now Adobe Audition 3.0 for $349USD)
can do it -jason