similar to: play alaw file with .wav extension

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

Interesting announcement today; http://www.crn.com.au/News/174872,microsoft-response-point-voip-gets-a-d irt-nap.aspx Microsoft this week revealed its intention to discontinue its Response Point, its small business VoIP system for companies with up to 50 employees. While not unexpected, the move is significant because Response Point was once a promising product in which Microsoft

hi all, We going to implement a music service which enable user to playback a song by dialing to a service number. The problem is that the amount of data is huge so we have to plae it on an different server which is connected to the asterisk's via internet. Does asterisk support playing a audio file from an resource locate in a remote host? Please help, Thanks, Quyps

Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [root at asterisk testing]# asterisk -rx "file convert testExtended2.wav testExtended2.alaw" Unable to open input file: testExtended2.wav [root at asterisk testing]# asterisk -rx "file convert testLong2.wav testLong2.alaw" Unable to open input file: testLong2.wav The wav-file is MONO,

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a

Hi all, I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1. It started ok with out cdr_addon_mysql.o. But when I put cdr_addon_mysql.o in to modules folder, it fail at start and the following out has been thrown: ---------- [root at localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk -f ${CLIARGS}

Hello list, I have the following in my voicemail.conf : [general] format=wav|wav49 When I receive a WAV-file from my Asterisk-server, I am unable to play the file... There is no player on my Fedora that wants to play the file. When I make my Asterisk-server unreachable, the incoming call goes to the voicemail of my ITSP. Don't know what system they use there but the mailing I get with the

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

hi, i try to convert wav file to gsm format.use following commands; sox net263-welcome.wav -r 8000 -g -c 1 net263-welcome.gsm resample -ql the file is located in /var/lib/asterisk/sounds/net263 but cant' play.do you know what's wrong? -- Executing Playback("SIP/1001-00000091", "net263/net263-welcome") -- <SIP/1001-00000091> Playing

Is there a way to get * to use wav files instead of gsm files for the voicemail, agents, and queues applications? Gsm does not give all the quality we would like to have, and we use no low bit rate codecs.

Hi, The function help.start() assume that the brower is netscape which is no longer the dominant brower in used. I am using now opera and other use mozilla or Microsoft IE. I can always type help.start(browser="opera"), but it is not convenience: I would like to be able to set the default brwsernone for all. But the real bug is that even after launching

Hi Developers, This is Viet at Sun Microsystems. Does openssh support Solaris 10 spacrc, sol 10 x86, and sol 10 x64. If so, could you please point me to the link in openssh web site, not sunfreeware.com, that says so and the link to download. I need an official link to declare to Sun that you do support sol 10. Thanks.

I have exactly the same problem. I encoded a BWF file to flac and then then decoded back to wav, using --keep-foreign-metadata, and I ended up with a bit-perfect copy of the original BWF. So I was fairly confident that the flac encoding is working correctly, the problem definitely appears to be with the flac players. Unfortunately it seems that the majority of flac players cannot play 24bit

Hi, I'm using glusterfs version 3.4.0 from gluster-epel[1]. Recently, I find out that there's a glusterfs version in base repo (3.4.0.36rhs). So, is it recommend to use that version instead of gluster-epel version? If yes, is there a guide to make the switch with no downtime? When run yum update glusterfs, I got the following error[2]. I found a guide[3]: > If you have replicated or

Hi guys, Being a true Linux geek, I've never been too much into sounds or sound files other than a few .mp3 songs I got. My question is pretty straightforward and simple. I see that the music format of choice for asterisk is .gsm. What can I use to listen to files in .gsm format and what is the most effective way of recording files into .gsm format? The last part of the question is

Hi all, Out IMAP client (Outlook and others) use login name like joedirt at domainname.com I need dovecot to convert it to joedirt-domainname_com since that is the local login name. Can someone give me a hint how to do that? Is there any simple way to do it or do I have to modify any source code? Thanks, Zoong

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

To install Oracle Database 10g, I need to increase the swap memory to 1004 MB. [root at server OpenBravo]# rpm -ivh oracle-xe-univ-10.2.0.1-1.0.i386.rpm Preparing... ########################################### [100%] This system does not meet the minimum requirements for swap space. Based on the amount of physical memory available on the system, Oracle Database 10g Express Edition

On Thu, May 14, 2020 at 11:31 AM John Hughes <john at calva.com> wrote: > On 14/05/2020 08:10, John Hughes wrote: > > I am having a problem with one of my callers who is using either g729 or > alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? > In fact from the sip debug it looks like it does, but then I get the > dreaded "channel.c:5630