Displaying 20 results from an estimated 3000 matches similar to: "Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x4 (ulaw)/0x8 (alaw)"
2011 Jun 28
1
Asked to transmit frame type slin, while native formats is 0x8 (alaw)
Asterisk 1.8.3.2
I have been getting this warning constantly on CLI in a call scenario where
I use local channels to connect SIP with PSTN.
I use callfile and local channel to first call a PSTN number and if
answered, use local channel to call SIP phone with music on hold enabled in
Dial string.
If I call PSTN from SIP directly or vice versa I don't see this warning
coming.
On SIP I have
2011 May 31
0
Dropping incompatible voice frame on DAHDI/i1/xxxxxxx of format slin since our native format has changed to 0x4 (ulaw)
Hey,
Sometime i am getting following messaged on asterisk CLI console just wondering what these messages are look like some codec related.
[May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our native format has changed to 0x4 (ulaw)
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2005 Mar 20
1
I cannot use G711 (ulaw|alaw)
Dear all,
I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to,
I got the following error message:
Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect
attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with
our capability 0xfe02.
I do not understand why because my Asterisk box load these codecs properly!
Does somebody
2006 Mar 15
1
dropping voice frame ulaw - slin?
Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice
frame on Local/[removed number]@context-5c3e,2 of format ulaw since our
native format has changed to slin
Can anyone provide an English translation of what this means?
The extension is a Polycom IP 501
The only allowed formats are g.711u
MOH is MP3 files (obvious)
All prompts have been re-recorded in .ul uLaw
2010 Oct 11
1
Unable to find a codec translation path from ulaw|h261 to slin
I'm doing some final check-outs before upgrading from 1.4.x to 1.6.x and I've
encountered a problem playing back a .wav file to an Ekiga client:
My dialplan looks like:
exten => 730,1,answer
exten => 730,n,playback(/home/phones/common/moh/moha/Sovereign)
exten => 730,n,hangup
Sovereign.wav is a .wav file that plays nicely on my 1.4 server.
Here is what the console displays:
2008 Mar 10
1
1.6.beta5 (format 0x40 (slin))
(alternative title - what did I do wrong? or suggestions to make this
work)
Thought I'd try 1.6 beta5 (and 1.4.18 didn't want to compile vpb
/usr/lib/gcc/i386-redhat-linux/4.1.2/../../../../include/c++/4.1.2/i386-redhat-linux/bits/gthr-default.h:48:
error: ? does not name a type )
1.6 did compile and almost works.
'cept it thinks the .gsm files are not played.
from
2010 Nov 19
0
help with annoying warning message: Asked to transmit frame type ulaw, while native formats is 0x2 (gsm) read/write = 0x2 (gsm)/0x2 (gsm)
Hi all,
i have a little problem to understand this warning message, it's annoying
and it cause a lot of spurious in the log files.
Im working with this scenario:
a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are
always routed to this.
a list of sip UAs that potentially can use any codec apart g729/g723.
I setup the asterisk to do as mediaproxy so directmedia=no and
2009 Jul 28
0
Asked to transmit frame type 256, while native formats is 0x4
Hi, sorry to bother u all, i have a trouble
when I call a did number forward to my asterisk server, the server told me:
[Jul 28 19:00:57] WARNING[28080]: chan_sip.c:3806 sip_write: Asked to
transmit frame type 256, while native formats is 0x4 (ulaw)(4) read/write =
0x4 (ulaw)(4)/0x4 (ulaw)(4)
[Jul 28 19:00:57] WARNING[28080]: chan_sip.c:3806 sip_write: Asked to
transmit frame type 4, while
2011 May 17
1
Name or service not known
Hi, my log is full of errors from this mobile user:
-- Registered SIP '0010106' at 212.93.97.135:7759
[2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804
handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms /
10000ms)
[2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245
ast_sockaddr_resolve: getaddrinfo("212.93.97.135:7759", "7759", ...):
Name
2017 Nov 22
3
Chan Local, Originate and slin
Hi all!
Asterisk 13.1.0 Ubuntu 16.04, all latest.
Can anybody explain this to me - I run Originate command from dialplan:
same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})
and I get crazy sound distortion in the conference, and I see that
transcoding takes place here:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin
2017 Nov 22
2
Chan Local, Originate and slin
Again - when Originate is run from dialplan, i get:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin at 8000)->(slin at 192000)
ReadTranscode: No
When it's made with a call file (no matter how a call file is created), I
see
NativeFormats: (slin)
WriteFormat: slin
ReadFormat: slin
WriteTranscode: No
ReadTranscode: No
Please
2005 Jun 15
0
Problem with slin
Hi all,
After upgrading to lates CVS head, I have problems using a IAXY device,
having slin problems:
Jun 15 18:59:31 NOTICE[8197]: channel.c:1475 ast_read: Dropping
incompatible voice frame on IAX2/lise-1 of format slin since our native
format has changed to ulaw
Because of that outside caller can't ear the callee on the IAXY.
Found somewhere that disabling transcode in asterisk.conf
2012 Feb 17
0
Calling SIP extension through Local/XXXX negotiates slin codec instead of gsm
I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga
on Fedora 16 x86_64 for my tests.
[root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf
2006 Apr 19
3
SLIN format
In sox terms is SLIN .ul (as in unsigned linear).
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net
Euro Tech News Blog http://eurotechnews.blogspot.com
2005 Sep 05
0
ooh323c h323_convertAsteriskCapToH323Cap Don't know how to deal with mode 0x40 (slin)
Hello,
I have the following setup:
(*)<--->IP<--->Micronet 5012 H.323 box <---> POTS <---> PBX (Alcatel OmniPCX)
Grand idea is to use the micronet's POTS interfaces to connect SIP
phones to the PBX and to the PSTN. I think i even managed my way in
the arcane and cryptic management interface of that appliance, but I
am stuck against theese messages:
-- Executing
2008 Nov 11
3
Use the NEW ulaw/alaw codecs (slower, but cleaner)
In Asterisk 1.6, there is an option to use the 'new g.711 algorithm'.
"Use the NEW ulaw/alaw codec's (slower, but cleaner)"
By slower does this mean more 'expensive', or does it instead mean that there will be more algorithmic latency? Both? Can anyone speak to the relative increases?
With regard to accuracy, can anyone speak to what kind of situation might
2010 Aug 20
2
codec_g729.so not work!
hi, all
i want to use g729 codec for set up a call. so i donwloaded the
so file from web site: http://asterisk.hosting.lv/#bin
and install it properly.
*CLI>
*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
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
2013 Feb 28
1
Transcoding issues with siren14
Sorry for a possible retransmit: the first was sent from an incorrect
email address.
I'm trying to use the Polycom SoundStation IP 7000 with Confbridge.
But the transcoding from siren14 to slin32 is via slin. First, it
seems odd that there's no transcoder directly to slin32 since anything
else will lower fidelity. But, more importantly, there is transcoding
from siren14 to slin16 and
2010 Aug 17
3
Convert wav-file to alaw-file
Hello list,
it seems that Asterisk is unable to convert a wav-file into an alaw-file :
[root at asterisk testing]# asterisk -rx "file convert testExtended2.wav
testExtended2.alaw"
Unable to open input file: testExtended2.wav
[root at asterisk testing]# asterisk -rx "file convert testLong2.wav
testLong2.alaw"
Unable to open input file: testLong2.wav
The wav-file is MONO,