similar to: Asterisk script to repeat dial of a number

Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081211/85bd0069/attachment.htm

Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? Asterisk Version 1.4.21.2 Executing [0872200189 at In:2] VoiceMail("SIP/voip-1fd034e0", "910|u") in new stack -- <SIP/voip-1fd034e0> Playing 'vm-theperson' (language

Hi, There is a long call setup time untill the call connects. How can I play a beep tone say every 4 seconds to the caller untill the call connects? Tx. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/38e17d3e/attachment.htm

Hi, Why does this warning occur and what are the implications of it? I'm concerned about calls never getting hung up.....! chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to call '2f197e56611061a678c13b881b2691a9 at 411.2.139.106'. Giving up. Tx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I've followed instructions of the book "AsteriskFutureOf TelephonySecEdit" on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . I can run aserisk fine from the non-root user. Please help Code Snippet: 1:

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

Say, I just picked this up on my messages! There are a whole host of these requests! Anyone know whow there people are? Is there a way to report them? Any suggestions as to how to block them? [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '"912" <sip:1 at 41.1.1.1>' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010]

Can I use grep ? Tried but not working. please help Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081212/c856a1f1/attachment.htm

Hi, I've tried the link http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error at the moment. Any other ideas most welcome. Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/c491a7b9/attachment.htm

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

http://lists.digium.com/pipermail/asterisk-users/2004-October/065540.html I saw this post about the M(x) option for the Dial command, but I could not find a reply questions posed here. I am wanting to pass the Zap channel that the original call came from to my macro embedded in the Dial command. I've tried to add arguments to the macro by using the syntax M(x,arg1), and I always get the

Hi all, Yesterday, in an attempt to take back my phone room, I pulled everything apart as far back as the demarc and rebuilt it. In the process of putting things back together I accidentally connected my incoming lines to my FXS ports and my phones to my FXO ports. I quickly realized the mistake I made and corrected things but not before one of my FXS modules was smoked by incoming ring voltage.

Let's suppose you have a PSTN line with multiple extensions. One of these extensions is connected to an Asterisk FXO port. For reasons that I won't go into, the "normal" configuration of the system is that when a call comes in on the PSTN line, Asterisk *does not answer* -- the other extensions do whatever they would do if there were no Asterisk system there. When I hear one

I came into this with my eyes wide open. I have read ABSOLUTELY EVERYTHING there is to be found on the net about avoiding echo problems BEFORE I even attempted to create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production

List, Im a Cisco certified Network guy with little telecom experience (BRI/PRI at the time) so please forgive my terminology. I am showing interest after the Network World SHSU October 4 article. We have 3 offices (Hub-Spoke T1 Frame relay to the remote offices(Data & voice on separate T)). Each office currently does their own thing for telecom. Our Main(HUB) office currently has 14 channels

Hi. I am using asterisk 1.4 (latest as of today) and zaptel 1.4 (latest as of today) and I cannot dial out using my 400P card with one fxs module and one fxo module. I am using kernel 2.6.24 and get the following log entries: [Feb 25 17:28:13] VERBOSE[25071] logger.c: -- Executing [s at macro-dialout-trunk:23] Dial("Zap/1-1", "ZAP/4/www411|300|wW") in new stack [Feb 25

Hi All, I am trying to achieve something pretty simple. I have a samba share and I would like all new directories and files within that share to have 775 permissions. My config for that share is below. Whenever I create a dir I get 755 permissions. I have set every force permission their is. Both configs below produced the same directory. I feel like this should be very easy and my settings are

I can't seem to get a Windows XP Media Center Edition system to see a very simple print share. [global] workgroup = HOME realm = SHAUN server string = Samba Server security = SHARE encrypt passwords = No log file = /var/log/samba/%m.log max log size = 50 dns proxy = No wins support = Yes valid users = shaun admin users = shaun read list = shaun write list = shaun printer admin

Hi All, I'm using samba 3.0.10-1 on RedHat Enterprise Linux 4. I'm authentication to our PDC using Winbind and also mounting our home directory share (from Windows) onto our main Linux server such that when a user logs on to the Linux server, their Windows home directory is mounted also. I'm using the domain admin account to mount the home directory share, which I feel is

Hi all, can anyone tell me how i can setup a recycle bin for my samba share? it will be a very very good for me.. please give me the steps or please point to any document.. Thank you in advance, Jerrynikky.