Displaying 20 results from an estimated 20000 matches similar to: "Codec ordering?"
2005 Feb 24
1
Problems with SIP codec selection
We've been using SIP with Asterisk for a couple of years now, and it's
generally worked fine. However we're now trying to use a more
complicated codec setup, and I've hit a problem with how codecs are
selected that I can't get around.
For a simple configuration:
XLite > GSM > Asterisk
where GSM is the _only_ codec selected on XLite, and in sip.conf we have:
2006 Mar 28
0
codec translation problem???
2005 Jul 03
0
H323 with GSM codec is not working
Hello,
I'm trying to use the GSM codec with Nufone H323 but it's not working.
Does somebody has some idea? Have I missed something?
Thanks!!
Celso Fassoni
Some additional info:
(I'm using CVS-HEAD - downloaded today)
monkey:~# cat /etc/asterisk/h323.conf
[general]
port = 1720
bindaddr = 192.168.0.100 ; this SHALL contain a single, valid
IP address for this machine
2006 Dec 07
1
Codec Selection in asterisk
I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and
they all are able to register and make calls with no problem . My voip
carrier supports gsm as well as ilbc .. Server takes calls from sip phones ,
does call recording in between and forwards to voip carrier . My problem is
that half of my softphones use ilbc and rest use gsm and my provider
supports both gsm as well as
2006 Feb 23
3
Codec order sent wrong from Asterisk
I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000.
The codec order on each one is the next:
SJPhone: GSM - iLBC - PCMA - PCMU
GXP2000: G729 - GSM - PCMA - PCMU
(I have a G729 license, so there's no problem with transcoding G729)
In my sip.conf, I've defined the following codec order:
disallow=all
allow=g729
allow=gsm
allow=g726
allow=alaw
allow=ulaw
And my
2004 Apr 08
0
Re: [Iaxclient-devel] codec negotiation ?
On Thu, 08 Apr 2004 10:14:09 -0400, Steve Kann wrote:
>Gary wrote:
>
>>I have noticed lack of codec negotiation with calls thru a registrated
>>asterisk box.
>>
>>No seen problems with outbound calls, (though I haven't specifically
>>tried it), but the problem exists inbound.
>>
>>Easiest method for testing this was ring in via a sip client set
2007 May 04
2
Asterisk Codec Translation Table
Hello list,
I have always though codec translation table is dircetly connected to system speed, utill i came across this:
in my lab, i have 2 boxes,
First box is an Intel Celeron 1.7 GHZ with 256M RAM:
show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw
2004 Oct 04
1
How to see CODEC which is in use?
How can I see which codec is in use during conversation. I can see (for
example) which codecs are negotiated before SIP connection, but I don't
know which is chosen:
12 headers, 12 lines
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Peer audio RTP is at port 217.10.79.30:15666
Found description format GSM
Found description format iLBC
2007 Apr 27
4
Unable to find a codec translation path from ilbc to ulaw
Hi!
As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:
SIP-phone <--iLBC--> Asterisk <---ulaw----> PSTN-Gateway
I get the following error:
"Unable to find a codec translation path from ilbc to ulaw"
Setup SIP-phone:
disallow=all
2010 Feb 08
3
High codec translation times on x64
Hi Users,
I was wondering if someone of you have the same thing on CentOS 64x?
asterisk01*CLI> core show translation
Translation times between formats (in microseconds) for one
second of data
Source Format (Rows) Destination Format (Columns)
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729
speex ilbc g726 g722 siren7 siren14 slin16
g723
2005 Feb 10
1
Codec passthrough patch for IAX
Hi there,
I had a problem, basically, I have 4 different types of end users
(gsm, ilbc, g729, ulaw). However, I only have one user with my DID provider.
My provider supports all 4 codecs. The issue is then: When an incoming call
comes in, a codec is negotiated (usually ULAW), later on, when the extension
is dialed, we'll see we're doing GSM, and thus transcode. Here's an example
2005 Sep 19
0
need a simply configuration for calling in/out to PSTN
Hi all,
I have configured Asterisk to call to PSTN phone from
our IP phone, But I am unable to call my IP phone from
a PSTN phone (If I called any number between 21494350
and 21494399, the card should route my call to my IP
phone, IF my configuration was correct). I have done
my research and gathered bits and pieces of
information (which are vague by the way) but still
cannot call my IP phone from
2008 Aug 09
1
how to know what codec is being used
Hi,
how would i know what codec is being utilized? currently i have set allow=ilbc disallow=all.
i unset all codecs on x-lite except ilbc.
i tried to make a call and look at the channel i see these. does this mean it is using ulaw? how about writetranscode? does that mean there is no transcoding happening on the call? call is going thru, rtp is also going thru. what i would like to know is does
2009 Oct 20
1
Is there a way to force a codec on an incoming sip uri call?
Hello,
I'd like to implement some public sip uri's that poeple can call into
and get an echo test. Is there a way to force a codec so that users
can test various codecs?
Something like:
echo-test at example.com (negotiates whatever codec, is there a way to
figure out what codec was negotiated and tell the user)
echo-test-g711 at example.com (forces g711)
echo-test-g729 at
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
2005 Mar 11
1
NuFone Configuration [problem]
Hello,
I am trying to configure the my asterisk box here with the following
**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xxxxxx
***extensions.conf:***
exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx@NuFone/${EXTEN}
exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN}
I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan.
2005 Feb 12
1
iax.conf config and iax based clients
Hi,
I am a newbie in asterisk. trying to configure firefly third party edition
to connect to aserisk 1.0.3 im able to authenticate but cannot dial
extensions. I have been reading the documentation cant seem to find the
correct configs. Attached the error message and configs. What am I
missing?
*CLI> Urgent handler
Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected
connect
2004 Jul 30
1
SIP connections do not hang up
Hi everybody,
I have strange problem:
I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone
using Zap Channel) using sipgate to a number in public network.
When I'm hanging up before the other person picked up the phone, the line is
not closed correctly.
The phone keeps on ringing until timeout (of Sipgate I assume) and it even
costs my money, if the other person
2012 Mar 21
0
AMR Codec with Asterisk 1.8.9.1
Hi All,
I would like to configure AMR codec in asterisk 1.8.9.1.
After lots of research i found "
http://sourceforge.net/projects/asterisk-amr/files/" thie link, and follow
steps to configure amr.
codec_amr.so successfully compiled and load in asterisk.
*> core show translation *
Translation times between formats (in microseconds) for one second
of data
Source
2010 Jul 05
1
Problems with ulaw/g729 translation
Dear Folks,
I'm running Asterisk 1.4.31 server, on an Ubuntu 9.10 system. My scenario is
simple: connection to the PSTN directly via SIP, using g729 codec, and
connection to the softphones (X-lite 3.0 build 56125) trought local network,
using ulaw codec.
Sometimes, I got messages like:
[Jul 1 15:26:16] WARNING[26483]: chan_sip.c:5514 process_sdp: Unsupported
SDP media type in offer: image