search for: slin32

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and slin16 to slin32. So why is slin used as the intermediate instead of slin16?

...;(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)->(slin32) alaw To slin44 : (alaw)->(slin)->(slin44) alaw...

...ven 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 15000 17250 1...

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...

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

There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play back

...unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * AST...

...ASTERISK-28538 - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp) * ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTER...

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

...unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * AST...

...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 audio RTP is a...

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 '24000' with id '11' == Created cached format with name 'slin24' == Registered '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 wit...

...D (Reported by Guido Falsi) * ASTERISK-23756 - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTER...

...D (Reported by Guido Falsi) * ASTERISK-23756 - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTER...

...unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * AST...