Displaying 20 results from an estimated 10000 matches similar to: "using iLBC"
2004 May 20
8
I have put iLBC at the top
I want use iLBC and have following in mind, please help me is it possible ?
ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM
SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM
1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC
because a lack of codec).
2. SIP incoming codec should be iLBC (snom is ALAW).
3. SIP outgoing codec should be iLBC /snom
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 May 19
1
Re: Grandstream ATA 286 and ilbc (Anton Krall)
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2005 Feb 18
3
MultiLine Sip Phones
Sorry Newbie asking everyones option.
I am setting up a couple of small asterisk phone systems for my work, I
started using some snom 190 and bt102 sip phones (the bt102 works really
well with iLBC), but the complaint from my workmates is there is no way
to see if other people are on there phone or not, or what lines are
being used.
The snom 190 only has 5 function keys, the snom 220 seems a bit
2011 May 08
3
Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello all,
I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
exercise. After setting up everything and trying to fix this problem,
I am still getting a 401 Unauthorized SIP message. So as of this
writing, I still cannot successfully REGISTER
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 May 12
2
GXP 2000 Conference Button and ILBC
Guys.
I just downloaded the recent firmware for GS GXP 2000 and I must say the
phone works great but... How do you make the conf button work?????? Anybody
done that?
Also, with great dissapointment I must ask, where is ILBC support? GS web
page mentions it and the manual says it supports it almost using bolds :)
soooooo where is it????
Any light on this?
2009 Jun 08
2
Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone,
i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM
300,320,360 Devices.
In the combination with asterisk and the patton, there are occuring some
strange behaviour, due to the calling and answering everything works
good, clear voice, great availability.
All the devices have to use ulaw, alaw and slinear is available but
never the first choice since i use my
2003 Apr 16
4
iLBC
i tried asterisk ilbc codec against kphone. when the call got
connected, i started to immediately get these kind of message to the
console:
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)?
WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of
2004 Sep 28
3
CODECs and sip.conf and voice quality
Group,
Just want to share with the group my recent findings regarding
CODECs/Vocoders and the effect it has had on voice quality and the
intermittent noise and breakup problem I have which I mentioned in a
previous emailing with the u-law CODEC. Calls again are placed through a
SIP phone to a TDM400P to the PSTN. A good reference on the reasoning
behind the selection of a CODEC was found in the
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
2008 Apr 13
1
compilation of asterisk 1.4.19 with ilbc already on system
I already have ilbc installed on my system. The files
are:
/usr/include/ilbc/iLBC_decode.h
/usr/include/ilbc/iLBC_define.h
/usr/include/ilbc/iLBC_encode.h
/usr/lib/libilbc.a
/usr/lib/libilbc.la
/usr/lib/libilbc.so -> libilbc.so.0.0.0
/usr/lib/libilbc.so.0 -> libilbc.so.0.0.0
/usr/lib/libilbc.so.0.0.0
However, if I do a "make" in asterisk-1.4.19, it will
not detect that libilbc.a
2006 Feb 13
1
iLBC issue: An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)
Hello all,
I've started implementing iLBC on some of the ATAs we have floating around
clients' homes, but I'm coming against this error message with most of them:
codec_ilbc.c: Huh? An ilbc frame that isn't a multiple of 50 bytes long
from RTP (38)?
The ATAs in question are various Grandstream models - the HT486 being the
predominant one. Looking at the list archives, it's
2020 Jun 13
5
Voice "broken" during calls
Am 13.06.2020 um 13:47 schrieb Michael Keuter:
Hi
> Try "sip show peer <peername>" for a phone.
So:
mobile phone:
bpi*CLI> sip show peer 0049177xxxxxxx
* Name : 0049177xxxxxxx
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : default
Record On feature : automon
2011 Jun 22
1
iLBC re-licence
Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?
https://sites.google.com/site/webrtc/ilbc-freeware
It seems to require that the Google iLBC licence document is on the
box, but that otherwise it is free-to use by all in any way (BSD
licence style). I believe that prior to that there was a requirement
to register
2005 Jul 28
2
SIP Debug
Using AMP, the configuration I have used to work fine with Broadvoice.
Now it gets a busy signal every time. I've checked "sip show
registry" and it says it's registered just fine. I've tried "sip
debug" and it shows calls coming in, but they always get a busy signal
& I can't tell why.
Here's a SIP Debug output:
Sip read:
INVITE
2004 Jan 26
3
X-Lite & Asterisk: Speex & iLBC not working?
This seems to have been reported before, but I've seen no resolution:
http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html
http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html
http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html
When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the
Asterisk server)
When forcing
2004 Oct 04
1
Will there be any support for iLBC in IAXClients soon?
Hello Folks,
I noticed that all of the IaxClient based softphones with exception of
Firefly only seem to have support for GSM but what about iLBC?
The quality is excellent with iLBC even on a dialup connection! Meanwhile
while the audio on GSM often sounds scratchy. Is anyone looking to
implement iLBC in an IaxClient based softphone soon?
Errol
Biz4Web Solutions Limited
2005 Sep 05
2
Speex or iLBC?
Hi kind developers,
I need select soon the best freeware VOIP codec, I see that all competitors
are using iLBC because of the separate packets management.
How speex behave in case of packets drop?
Why other choice all iLBC?
Thank you for any kind answer.
Best regards.
-------------------------------------
Roberto Della Pasqua
Http: www.dellapasqua.com
Email/Msn: roberto@dellapasqua.com
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