search for: slin44

...Set("SIP/sipuser-00000001", "ABA=999999999") in new stack -- Executing [36225 at default:2] BackGround("SIP/sipuser-00000001", "telbank/999999999/en/thetimeis") in new stack -- <SIP/sipuser-00000001> Playing 'telbank/999999999/en/thetimeis.slin44' (language 'en') [Aug 14 14:22:45] NOTICE[8690]: channel.c:4202 __ast_read: Dropping incompatible voice frame on SIP/sipuser-00000001 of format gsm since our native format has c hanged to (ulaw) -- Executing [36225 at default:3] SayUnixTime("SIP/sipuser-00000001", "...

...as combining and playing using playback/background in 1.4.X. Now that I have moved on to the 10.0 set, I understand that I can replace my 8 Khz mono files with virtually unlimited Khz mono files (still no stereo, but a quantum leap forward). I've played with this and get good throughputs using SLIN44 formats on SIP. The 2 questions I have are: 1. Is Slin44 the format I should be settling on or has someone found a combination they find preferable? 2. While the SIP connections sound good, I still have to "talk" through OBI110 DADHI devices and other UUCM type connections -...

...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) alaw To slin96 : (alaw)->(slin)->(slin96) alaw To slin192 : (alaw)->(slin)->(slin192)

...sical 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 15000 17250 17000...

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 1...

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

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 =======================================

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

...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 audio RTP is at port...

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

...egistered 'audio' codec 'slin' at sample rate '32000' with id '12' == Created cached format with name 'slin32' == Registered '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 wit...