Displaying 20 results from an estimated 25 matches for "testlaw".
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testlab
2019 Jul 05
2
Asterisk and Linphone
I have no speex translation
ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw
ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000 17000 17000 15000 15000 17250 15000
alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000
17000 17000 17000 17000 17000 15000 15000 17250 15000
gsm 15000 15000...
2013 Mar 21
2
Allow/Disallow
Hello Everyone,
I have disallow=all and allow=g729 set in sip.conf however, it seems
that asterisk still thinks it support other codecs:
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw). How
can I disable gsm,ulaw,alaw.....
Thanks in Advance,
Nick.
2012 Mar 21
0
AMR Codec with Asterisk 1.8.9.1
...ore 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 amr siren14 slin16 g719 speex16 siren7 testlaw
g723 - - - - - - - - -
- - - - - - - - - - -
gsm - - 1001 1001 - - 1000 - -
10999 - - - *9998* - - - - - 2000
ulaw - 30...
2014 Feb 11
0
g726 transcoding
...alaw To g726aal2 : No Translation Path
alaw To g722 : No Translation Path
alaw To slin16 : (alaw)->(slin)->(slin16)
alaw To siren7 : No Translation Path
alaw To siren14 : No Translation Path
alaw To testlaw : (alaw)->(slin)->(testlaw)
alaw To g719 : No Translation Path
alaw To speex32 : (alaw)->(slin)->(slin32)->(speex32)
alaw To slin12 : (alaw)->(slin)->(slin12)
alaw To slin24 : (alaw)->(slin)->(slin...
2019 Jul 05
4
Asterisk and Linphone
Hi all - I am using asterisk 13.27.0 with Linphone.
I turned off all codes on linphone except the one I want to try. For
example:
opus and speex (so only one enabled at a time).
Then did this same on asterisk for the linphone extension.
disallow=all
allow=speex
(for example).
Then I place my call and the call fails. if I enable something like gsm,
ulaw, alaw the call works fine. Why does the
2019 Jul 08
3
opus codec
...t;(ilbc at 8000)
opus:48000 To g722:16000 : (opus at 48000)->(slin at 48000
)->(slin at 16000)->(g722 at 16000)
opus:48000 To siren7:16000 : No Translation Path
opus:48000 To siren14:32000 : No Translation Path
opus:48000 To testlaw:8000 : (opus at 48000)->(slin at 48000
)->(slin at 8000)->(testlaw at 8000)
opus:48000 To g719:48000 : No Translation Path
opus:48000 To none:8000 : No Translation Path
opus:48000 To silk:8000 : No Translation Path
opus:...
2012 Nov 21
1
core show translation - difference in Asterisk Versions
...on Asterisk 11 (VM, Cloud or even physical machine). Is it slin?, adding
this overhead or there is something I am overlooking?.
*
*
*Asterisk 11.0.1 => core show translation **(in microseconds)*
*gsm ulaw alaw g726 adpcm slin lpc10 g729 speex speex16
ilbc g726aal2 g722 slin16 testlaw speex32 slin12 slin24 slin32 slin44
slin48 slin96 slin192*
*gsm *- 15000 *15000 *15000 15000 9000 15000 15000 *15000 *23000
15000 15000 17250 17000 15000 23000 17000 17000 17000 17000
17000 17000 17000
*ulaw *15000 - 9150 15000 15000 9000 15000 15000 15000...
2020 Jun 13
5
Voice "broken" during calls
...ddr->IP : (null)
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username:
SIP Options : (none)
Codecs :
(alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk)
Auto-Framing : No
Status : UNKNOWN
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Refuse
Sess-Refresh : uac
Sess-Expires : 1800 secs
Min-Sess : 90...
2020 Jun 13
0
Voice "broken" during calls
...Defaddr->IP : (null)
> Prim.Transp. : UDP
> Allowed.Trsp : UDP
> Def. Username:
> SIP Options : (none)
> Codecs :
> (alaw|ulaw|ilbc|g729|g723|gsm|amr|amrwb|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk)
> Auto-Framing : No
> Status : UNKNOWN
> Useragent :
> Reg. Contact :
> Qualify Freq : 60000 ms
> Keepalive : 0 ms
> Sess-Timers : Refuse
> Sess-Refresh : uac
> Sess-Expi...
2020 Jun 13
3
Voice "broken" during calls
Am 13.06.2020 09:30, schrieb Luca Bertoncello:
Hi again (again)
I noticed right now another strange detail...
I made a call using my mobile phone (connected to the Asterisk). The
quality was top...
Maybe is the problem in a codec used from our phones at homes?
Could someone suggest me how to check the codec used by my mobile phone
and the codec used by the phones at home?
Thanks
Luca
2017 Apr 19
2
Asterisk 1.8.32.3 : no video (h.264)
...<< 44) (0x100000000000) audio unknown
(unknown)
35184372088832 (1 << 45) (0x200000000000) audio unknown
(unknown)
70368744177664 (1 << 46) (0x400000000000) audio unknown
(unknown)
140737488355328 (1 << 47) (0x800000000000) audio testlaw
(G.711 test-law)
281474976710656 (1 << 48) (0x1000000000000) video unknown
(unknown)
562949953421312 (1 << 49) (0x2000000000000) video unknown
(unknown)
1125899906842624 (1 << 50) (0x4000000000000) video unknown
(unknown)
2251799813685248...
2016 Dec 10
6
failing to start asterisk on centos7
...te '16000' with id '24'
== Created cached format with name 'siren7'
== Registered 'audio' codec 'siren14' at sample rate '32000' with id '25'
== Created cached format with name 'siren14'
== Registered 'audio' codec 'testlaw' at sample rate '8000' with id '26'
== Created cached format with name 'testlaw'
== Registered 'audio' codec 'g719' at sample rate '48000' with id '27'
== Created cached format with name 'g719'
== Registered 'audio'...
2014 Aug 25
0
WebRTC / Rejecting secure audio stream errors
...ion format PCMA for ID 8
Found audio description format telephone-event for ID 101
-[Aug 25 14:59:29] WARNING[3938][C-00000003]: chan_sip.c:10535
process_sdp: Rejecting secure audio stream without encryption details:
audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101
+Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
+Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
+Peer audio RTP is at port 195.8.117.59:51390
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
...> (unknown)
>> 35184372088832 (1 << 45) (0x200000000000) audio unknown
>> (unknown)
>> 70368744177664 (1 << 46) (0x400000000000) audio unknown
>> (unknown)
>> 140737488355328 (1 << 47) (0x800000000000) audio testlaw
>> (G.711 test-law)
>> 281474976710656 (1 << 48) (0x1000000000000) video unknown
>> (unknown)
>> 562949953421312 (1 << 49) (0x2000000000000) video unknown
>> (unknown)
>> 1125899906842624 (1 << 50) (0x4000000000000...
2011 Oct 11
0
Asterisk 1.8.7 and VoiceMailMain
...2011-10-11 08:06 msg0008.wav
-rw-rw---- 1 asterisk asterisk 5715 2011-10-11 08:06 msg0008.WAV
Codec negotiation:
Capabilities: us - 0x80030c7fffff
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719),
peer - audio=0xc (ulaw|alaw)/video=0x380000 (h263|h263p|h264)/text=0x0
(nothing), combined - 0x38000c (ulaw|alaw|h263|h263p|h264)
In asterisk.conf we even activate
transcode_via_sln = yes ;Build transcode paths via SLINEAR,instead of
directly.
Why is Asterisk trying to read messages in...
2012 Jul 12
1
Asterisk with OpenBTS and mobile phone
...ransp. : UDP
> Allowed.Trsp : UDP
> Def. Username: 6201
> SIP Options : (none)
> Codecs : 0x80030c7fffff
> (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719)
> Codec Order : (none)
> Auto-Framing : No
> Status : Unmonitored
> Useragent :
> Reg. Contact :
> Qualify Freq : 60000 ms
> Sess-Timers : Accept
> Sess-Refresh : uas
> Sess-Expires : 1800 secs
>...
2010 Nov 01
0
Queue Group not forwaring calls to agents
...: 500
Timer B : 32000
ToHost : aethercommunications.com
Addr->IP : 173.203.87.134:5060
Defaddr->IP : 97.74.144.17:5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 9012211610
SIP Options : (none)
Codecs : 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
Codec Order : (none)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Us...
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
...000) audio unknown
> (unknown)
> 35184372088832 (1 << 45) (0x200000000000) audio unknown
> (unknown)
> 70368744177664 (1 << 46) (0x400000000000) audio unknown
> (unknown)
> 140737488355328 (1 << 47) (0x800000000000) audio testlaw
> (G.711 test-law)
> 281474976710656 (1 << 48) (0x1000000000000) video unknown
> (unknown)
> 562949953421312 (1 << 49) (0x2000000000000) video unknown
> (unknown)
> 1125899906842624 (1 << 50) (0x4000000000000) video unknown
> (u...
2015 Apr 28
0
hi list need your help
...S) : '7cvtd9ihs2e8.invalid'
[Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:98
ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
Audio is at 16476
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.196.158.205:1466:
INVITE sip:0momhddj at 7cvtd9ihs2e8.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 77.91.132.9:5060;branch=z9hG4bK3f293d79;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 77...
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua