Displaying 20 results from an estimated 22 matches for "g719".
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1719
2014 Jan 23
1
mixmonitor extension
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
--
---------------------------------------
Marek Cervenka
=======================================
2012 Mar 21
0
AMR Codec with Asterisk 1.8.9.1
...asterisk.
*> 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 amr siren14 slin16 g719 speex16 siren7 testlaw
g723 - - - - - - - - -
- - - - - - - - - - -
gsm - - 1001 1001 - - 1000 - -
10999 - - - *9998* - - - - - 2...
2020 Jun 13
5
Voice "broken" during calls
...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 secs...
2019 Jul 05
2
Asterisk and Linphone
...To ilbc:8000 : No Translation Path
speex:8000 To g722:16000 : No Translation Path
speex:8000 To siren7:16000 : No Translation Path
speex:8000 To siren14:32000 : No Translation Path
speex:8000 To testlaw:8000 : No Translation Path
speex:8000 To g719:48000 : No Translation Path
speex:8000 To opus:48000 : No Translation Path
speex:8000 To none:8000 : No Translation Path
speex:8000 To silk:8000 : No Translation Path
speex:8000 To silk:12000 : No Translation Path
speex:8000 To silk:1600...
2020 Jun 13
0
Voice "broken" during calls
...->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 :...
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
2020 Jun 13
0
Voice "broken" during calls
..."sip show peer <peername>" for a phone.
> bpi*CLI> sip show peer 0049177xxxxxxx
> 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|t
> estlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk
> |silk|silk)
That strikes me as somewhat unlikely.
> bpi*CLI> sip show peer 0049351xxxxxxx
> Codecs : (alaw|ulaw|ilbc|g729|g723|gsm)
That looks a little more standard.
Regards,
Antony.
--
I just got a new...
2014 Feb 11
0
g726 transcoding
...o 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)->(slin24)
alaw To slin32 : (alaw)->(slin)->(sl...
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)
...< 29) (0x20000000) text unknown
(unknown)
1073741824 (1 << 30) (0x40000000) (unk) unknown
(unknown)
2147483648 (1 << 31) (0x80000000) (unk) unknown
(unknown)
4294967296 (1 << 32) (0x100000000) audio g719
(ITU G.719)
8589934592 (1 << 33) (0x200000000) audio speex16
(SpeeX 16khz)
17179869184 (1 << 34) (0x400000000) audio unknown
(unknown)
34359738368 (1 << 35) (0x800000000) audio unknown
(unknown)
68719476736 (...
2014 Dec 11
0
PJSIP configuration question
...00
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:119 speex/32000
a=rtpmap:107 opus/48000/2
a=fmtp:107 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:96 SILK/8000
a=fmtp:96 maxaveragebitrate=10000
a=fmtp:96 u...
2016 Dec 10
6
failing to start asterisk on centos7
...te '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' codec 'opus' at sample rate '48000' with id '28'
== Created cached format with name 'opus'
== Registered 'image' c...
2017 Apr 20
2
Asterisk 1.8.32.3 : no video (h.264)
...; (unknown)
>> 1073741824 (1 << 30) (0x40000000) (unk) unknown
>> (unknown)
>> 2147483648 (1 << 31) (0x80000000) (unk) unknown
>> (unknown)
>> 4294967296 (1 << 32) (0x100000000) audio g719 (ITU
>> G.719)
>> 8589934592 (1 << 33) (0x200000000) audio speex16
>> (SpeeX 16khz)
>> 17179869184 (1 << 34) (0x400000000) audio unknown
>> (unknown)
>> 34359738368 (1 << 35) (0x80000000...
2011 Oct 11
0
Asterisk 1.8.7 and VoiceMailMain
...-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 slin...
2019 Jul 08
3
opus codec
...(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:48000 To silk:12000 : No Translation Path
opus:48000 To silk:16000 : No Translation Path...
2012 Jul 12
1
Asterisk with OpenBTS and mobile phone
...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
>...
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
...) text unknown
> (unknown)
> 1073741824 (1 << 30) (0x40000000) (unk) unknown
> (unknown)
> 2147483648 (1 << 31) (0x80000000) (unk) unknown
> (unknown)
> 4294967296 (1 << 32) (0x100000000) audio g719
> (ITU G.719)
> 8589934592 (1 << 33) (0x200000000) audio speex16
> (SpeeX 16khz)
> 17179869184 (1 << 34) (0x400000000) audio unknown
> (unknown)
> 34359738368 (1 << 35) (0x800000000) audio unknown
> (un...
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2019 Apr 17
2
IPv6 transport results in ICE with only IPv6 candidates
...6/16000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=rtpmap:107 opus/48000/2
a=ptime:20
a=maxptime:20
a=sendrecv
m=video 11188 RTP/AVP 31 34 103 99 104 100 108
a=ice-ufrag:28e29ae16a4cc8d54ebf03cf56010f1a
a=ice-pwd:57ec01cf24ac689d2fb459bd0411d2b6
a=candidate:H67c355d6 1 UDP 2130706431 fd31:aeb1:48df::2 11188 typ host
a=candidate:Hc1e19c54 1...
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