search for: slin12

Displaying 10 results from an estimated 10 matches for "slin12".

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2013 Jun 10
3
no silk translation ?
Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make
2014 Feb 11
0
g726 transcoding
...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)->(slin32) alaw To slin44 : (alaw)->(slin)->(slin44) alaw To slin48 : (alaw)->(slin)->(slin48)...
2012 Nov 21
1
core show translation - difference in Asterisk Versions
...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 23000 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 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
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 =======================================
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
...for ID 3 Found audio description 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...
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
...== Registered 'audio' codec 'slin' at sample rate '8000' with id '8' == Created cached format with name 'slin' == Registered 'audio' codec 'slin' at sample rate '12000' with id '9' == Created cached format with name 'slin12' == Registered 'audio' codec 'slin' at sample rate '16000' with id '10' == Created cached format with name 'slin16' == Registered 'audio' codec 'slin' at sample rate '24000' with id '11' == Created cached format wit...