similar to: SLIN format

Displaying 20 results from an estimated 5000 matches similar to: "SLIN format"

2008 Jan 15
3
Meetme recording
Hello, Is there a way to change the format from the default? 'r' - Record conference (records as ${MEETME_RECORDINGFILE} using format ${MEETME_RECORDINGFORMAT}). Default filename is meetme-conf-rec-${CONFNO}-${UNIQUEID} and the default format is wav. - requires chan_zap.so Many thanks ******************************************************************** This email and any attachments
2023 Jul 07
1
Asterisk Release 20.3.1
The Asterisk Development Team would like to announce security release Asterisk 20.3.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release:
2023 Jul 07
1
Asterisk Release 20.3.1
The Asterisk Development Team would like to announce security release Asterisk 20.3.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.3.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release:
2004 Dec 17
6
Realtime and PostgreSQL
Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing
2005 Jun 15
1
app_dial.c:977 dial_exec_full: Unable to create channel of type 'Zap' (cause 0)
Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give
2011 Sep 30
1
Core show translation > 4000ms
Hi list, we have 2 asterisk boxes in VM (kvm) on 2 different Dell servers, one is Lenny kernel 2.6.26 asterisk 1.6.2.20, the second CentOS 2.6.18 asterisk 1.4.36 (Elastix). Both 64bits, no hardware involved, dahdi on both machines for meetme timing. Doing core show translation give on the Lenny server Translation times between formats (in microseconds) for one second of data
2006 Jan 04
2
suddenly iax calls don't work anymore
Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial("Zap/2-1",
2012 Feb 22
1
How does format_mp3 work?
Hi I was using the Playback application to play an MP3 file after compiling and installing asterisk 1.8.7.0 with format_mp3 and it seems to me that asterisk is transcoding the file to an slin on the fly rather than playing the mp3 itself. Is this what it does? Also, does this mean I might as well change the format of MP3s to WAV seeing as I'm used to doing that anyway? Thanks Ish --
2006 Apr 06
1
Suggested MeetMe feature: 'i' without review.
I recently setup app_meetme with the 'i' option. My boss wants users to say their name and go directly into the conference instead of reviewing the recording. If anyone else is interested in this behavior becoming an option, has a suggestion what letter to use as the option (I was thinking 'i' -- with review and 'I' -- without review), or anything else, I'd appreciate
2009 Jun 08
2
Snom, Asterisk and Patton SN1400 - sending bye instead of hold
Hey Everyone, i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM 300,320,360 Devices. In the combination with asterisk and the patton, there are occuring some strange behaviour, due to the calling and answering everything works good, clear voice, great availability. All the devices have to use ulaw, alaw and slinear is available but never the first choice since i use my
2006 Jun 01
4
G729, voicemail, no codec_g729
I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the
2004 Sep 06
1
Problem Loading asterisk_oh323-0.6.3b eith last *cvs...
Hello I?ve just install last cvs version (Mon Sep 6) of Asterisk with asterisk-oh323-0.6.3b and pwlib-v1_6_6-src.tar.gz, openh323-v1_13_5-src.tar.gz and . this is the error loading asterisk with chan_oh323 module:: [cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_manager.so] => (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found
2004 Dec 15
5
How "expensive" are the different codecs? (Regarding CPU time)
Hi! The encoding, decoding and recoding cost cpu time, that's sure. But does this time differs much depending on the used codec? Is - for example - a G729 faster than a GSM codec? Bye! Michael
2007 Jun 16
1
Convert or listen to .sln file
Hi, How do I listen to .sln audio file or convert it to some format that can listened to? Sox does not seems to support .sln as an input file. Thanks. - Andrew
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 =======================================
2017 Nov 22
3
Chan Local, Originate and slin
Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin
2017 Nov 22
2
Chan Local, Originate and slin
Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When it's made with a call file (no matter how a call file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Please
2006 Mar 15
1
dropping voice frame ulaw - slin?
Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice frame on Local/[removed number]@context-5c3e,2 of format ulaw since our native format has changed to slin Can anyone provide an English translation of what this means? The extension is a Polycom IP 501 The only allowed formats are g.711u MOH is MP3 files (obvious) All prompts have been re-recorded in .ul uLaw
2017 Nov 22
2
Chan Local, Originate and slin
On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of
2006 Apr 04
1
Can't recieve Fax: No carrier detected - Ast erisk + iaxmodem + Hylafaxv --- sorry.wrong log.
> -- Format for call is ulaw Try the slin codec, I didn't have good results until I used slin.