Displaying 14 results from an estimated 14 matches for "slin48".
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slin44
2014 Feb 11
0
g726 transcoding
...gt;(speex32)
alaw To slin12 : (alaw)->(slin)->(slin12)
alaw To slin24 : (alaw)->(slin)->(slin24)
alaw To slin32 : (alaw)->(slin)->(slin32)
alaw To slin44 : (alaw)->(slin)->(slin44)
alaw To slin48 : (alaw)->(slin)->(slin48)
alaw To slin96 : (alaw)->(slin)->(slin96)
alaw To slin192 : (alaw)->(slin)->(slin192)
2012 Nov 21
1
core show translation - difference in Asterisk Versions
...achine). 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 23000
15000 15000 17250 17000 15000...
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 15...
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
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
=======================================
2016 Nov 23
0
Asterisk 14.2.0 Now Available
...t behaving badly - regression
(Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state (Reported by Joshua
Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
payload types. (Reported by Alexander Traud)
* ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when
publishing, in publisher_client_send at
res_pjsip_outbou...
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
2014 Dec 11
0
PJSIP configuration question
...tion format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (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|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 66.241....
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
2016 Dec 10
6
failing to start asterisk on centos7
...egistered 'audio' codec 'slin' at sample rate '44100' with id '13'
== Created cached format with name 'slin44'
== Registered 'audio' codec 'slin' at sample rate '48000' with id '14'
== Created cached format with name 'slin48'
== Registered 'audio' codec 'slin' at sample rate '96000' with id '15'
== Created cached format with name 'slin96'
== Registered 'audio' codec 'slin' at sample rate '192000' with id '16'
== Created cached format wi...
2017 Oct 03
0
Asterisk 15.0.0 Now Available
...adly -
regression
(Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state
(Reported by
Joshua Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the
SDP Media Attributes When SLIN48 Codec Is Used
(Reported
by Frankie Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32
dynamic payload types.
(Reported by Alexander Traud)
* ASTERISK-26546 - mips64el and x32 - undefined reference to
symbol 'dlopen@@GLIBC_2.2'
(Reported by...
2017 Aug 02
2
Asterisk 15.0.0-beta1 Now Available
...adly -
regression
(Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state
(Reported by
Joshua Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the
SDP Media Attributes When SLIN48 Codec Is Used
(Reported
by Frankie Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32
dynamic payload types.
(Reported by Alexander Traud)
* ASTERISK-26546 - mips64el and x32 - undefined reference to
symbol 'dlopen@@GLIBC_2.2'
(Reported by...
2016 Jul 27
3
Asterisk 14.0.0-beta1 Now Available
...ing (Reported by
Nuno Borges)
* ASTERISK-24516 - [patch]Asterisk segfaults when playing back
voicemail under high concurrency with an IMAP backend (Reported
by David Duncan Ross Palmer)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
Chin)
* ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
by xrobau)
* ASTERISK-24542 - [patch]Failure showing codecs via 'core show
channeltype <tech>' (Reported by snuffy)
* ASTERISK-24469 - Security Vulnerability...
2019 Dec 24
0
Certified Asterisk 16.3-cert1 Now Available
...owse/ASTERISK-26549>] -
app_dial: When PickupChan() is used some channels may have incorrect device
state
(Reported by Joshua C. Colp)
- [ASTERISK-24274
<https://issues.asterisk.org/jira/browse/ASTERISK-24274>] -
[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48
Codec Is Used
(Reported by Frankie Chin)
- [ASTERISK-26311
<https://issues.asterisk.org/jira/browse/ASTERISK-26311>] -
[patch] rtp_engine: Allow more than 32 dynamic payload types.
(Reported by Alexander Traud)
- [ASTERISK-26546
<https://issues.asterisk.org/jira/browse/ASTER...