search for: slinear

...g the problem, I found that for some reason the tone >>>>>generator, which uses 16-bit Signed Linear PCM, was still being >>>>>allocated and playtones_generator (indications.c) was still getting >>>>>called, regardless that the Sipura doesn't take SLINEAR data (in my >>>>>case, it accepts G711u). So, I ended up adding an if conditional to >>>>>the beginning of the playtones_alloc function (indications.c) to check >>>>>if SLINEAR was supported by the channel, and if not, return 0 (which, >>>&...

...istortion. I know this must be covered in the archives, but I can't find a reference. I am using the Asterisk demo configurations. configuration is as follows: ; ; Linux Telephony Interface ; ; Configuration file ; [interfaces] ;mode=immediate ;mode=fxo mode=dialtone ;format=g723.1 ;format=slinear format=ulaw silencesupression=no ; ; List all devices we can use. Contexts may also be specified ; context=local context=default ; txgain=100% rxgain=100% device => /dev/phone0

...8]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Thanks! -- Kristian Kielhofner

...sk 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 asterisk in europe. (slinear is available for debugging supposes) But if a calls comes from or go to the SN1400 and someone tries to HOLD a call, the snoms are sending bye instead of hold, Asterisk plays his MOH until the bye reveives, the...

...ger.c: -- Called 1000 Aug 10 17:43:44 DEBUG[1810] channel.c: Driver for channel 'mISDN/1-1' does not support indication 3, emulating it Aug 10 17:43:44 DEBUG[1810] channel.c: Scheduling timer at 160 sample intervals Aug 10 17:43:44 DEBUG[1810] channel.c: Building translator from alaw to SLINEAR for spies on channel mISDN/1-1 Aug 10 17:43:44 DEBUG[1810] channel.c: Generator got voice, switching to phase locked mode Aug 10 17:43:44 DEBUG[1810] channel.c: Scheduling timer at 0 sample intervals Aug 10 17:43:44 WARNING[1810] indications.c: Can't generate that much data! Aug 10 17:43:44 DEB...

In sox terms is SLIN .ul (as in unsigned linear). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog http://eurotechnews.blogspot.com

...ry log I can think of and there are no errors. Thanks, Steve Iaxmodem config: device /dev/ttyIAX0 owner uucp:uucp port 4570 server 127.0.0.1 peername iaxmodem0 secret itsasecret cidname Fax1 cidnumber 8005551212 codec slinear ttyIAX0 Config: FAXNumber: +1.800.385.7032 LongDistancePrefix: 1 InternationalPrefix: 011 DialStringRules: etc/dialrules ServerTracing: 1 SessionTracing: 11 RecvFileMode: 0600 LogFileMode: 0600 DeviceMode: 0600 RingsBefor...

...e of sending one single Fax. Actually when I try to send a Fax, the call is established between my * server and the remote Fax but after 30 secs Asterisk disconnects the call and Hylafax reports NO CARRIER DETECTED. Tried playing around with a few parameters such as no echocancellation, alaw (also slinear) codec, faxdetection =incoming in zaptel but with no luck. Regards, Jose Limeres -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070415/6df5f455/attachment.htm

Hi, I have been experimenting with Asterisk and the linejack card and have discovered the follow: 1) The linejack in FXS mode works well with asterisk as long as the format in the phone.conf is set to ulaw. With mode set to slinear, gsm recorded messages played by voicemail appear to be slightly choppy (sounds like a click at the end of each frame). 2) Using the configuration above, the linejack works well with the echo test. 3) Voicemail records smoothly and clearly as long as the format in voicemail.conf is set to gsm. Us...

I have a Quicknet Linejack in /dev/phone0. My phone.conf is: [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device => /dev/phone0 Only I can mark 7 digits, soon asterisk tries dial automatically. I cannot mark 8 or more digits. 6 or less digits work ok. -------------- next part -------------- An HTML attachment was scrubbed... URL: h...

I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071113/dea0f584/attachment.htm

...that is either encoding or decoding. Which means each *leg* of a call, if it is being transcoded, whether that is a single caller in a multi-caller (eg, 2 people or more) or even an app. So if both people in a call are sending G.729 encoded data, and your app decodes the *mixed* G.729 into ulaw (or slinear or any other decoded format it outputs) requiring a single instance of the decoder, then you need a single license. Multiple simultaneous calls working exactly like that each need a single license, #licenses = #calls. But if your app decodes both G.729 legs into ulaw (or other working format) data...

I need help creating a channel for my phone device (Quicknet PhoneJack). I have installed and loaded the driver and phone devices listen in /dev (phone0 - phone15). [phone.conf] mode=dialtone format=slinear device => /dev/phone0 fxoks=2 ;Quicknet PhoneJack [extensions.conf] ... exten=>_NXXNXXXXXX,1,Dial,Phone/phone0 ... When I try to make a call, I get the following output: Executing Dial("Phone/phone0", "Phone/phone0) in new stack NOTICE[262159]: File app_dial.c, Line 476 (d...

I have noticed that when I try to connect multiple G729 VoIP devices into a MeetMe conference that I can only add up to the number of G729 licenses I have. Now I would think that because all the devices are G729, this wouldn't be the case and the only license that would ever be used would be if a non G729 device or Zap channel was a part of the Meetme conference. This is apparently note the

...t sipgate.co.uk>;tag=as30eb9dd1' -- SIP/sipgate-0000014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register => SIP-ID:SIP-Password at sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=pr...

...B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the phone? _________________________________________________________________ Help STOP SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail

...gits or providing ; any dialtone (this is the immediate mode, the default). Also, you ; can set the mode to "fxo" if you have a linejack to make it operate ; properly. ; mode=immediate ;mode=dialtone ;mode=fxo ; ; You can decide which format to use by default, "g723.1" or "slinear". ; XXX Be careful, sometimes the card causes kernel panics when running ; in signed linear mode for some reason... XXX ; ;format=slinear format=g723.1 ; ; And set the echo cancellation to "off", "low", "medium", and "high". ; This is not supported on al...

...is what I tried: (in each case asterisk invoked with '-vvv -C name_of_config_file) 1. temporary fresh install of 1.2.40 using sample configs; in dialplan (uncommented extension ext. 1265); in 'phone.conf' uncommented the following: [interfaces] mode=dialtone format=slinear echocancel=medium context=local txgain=100% rxgain=1.0 device => /dev/phone0 -- dialplan works, ext. 1265 rings, has two-way audio, call progress tones heard, sends dtmf, can dial out, but has no dialtone. 2. temporary 'update' (bininstall) install of 1.2.40...

...not routing to my server. When I execute "sip show registry", its not displaying anything. Here I am giving my configuration details: My sip.conf file contents: [general] port = 5060 bindaddr = 0.0.0.0 qualify=no disable=all allow=alaw allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes [250] type=friend username=250 secret=danny callerid="Danny" host=dynamic context=demo register => 100xxxx:password@sipgate.co.uk/100xxxx [sipgate4] type=friend disallow=all allow=alaw allow=ulaw fromuser=100xxxx authuser=100xxxx secret=password username=100xxxx host=s...

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first