similar to: Problem getting Voice Contract script to work

This may sound kind of crazy and I maybe missing something. But are you placing the call on hold so you can hear the hold music. This may not be the case but you may have to place the call on hold to here the music. Also you mentioned sound, you do not need a sound card in the asterisk box to use this hold music feature. Hope this helps. -----Original Message----- From:

Hi, I've just done up a quick proggy to show me the status of my IAX peers from my windows box. It plugs into the simple manager proxy. You can see more information (including a screenshot) at: http://www.sineapps.com/news.php?rssid=384 You can download it directly from: http://www.sineapps.com/down/IAXPeers.zip Could you please have a look and let me know your thoughts. -- Cheers,

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

We've got some replies to questions online about Astricon and we now have a mirror available at: http://astricon.voctel.com/news.php If anyone has any comments about Astricon, please forward them to me and I will put them up on the site so that all the people who didn't go can read them. Cheers, Matt Riddell http://www.sineapps.com/news.php (Daily Asterisk News - html)

Hi, Just thought I'd let everyone know of our latest interview - this time with Mark Spencer - the creator of Asterisk. -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report "Not Response". I modified the audio level in zapata.conf and after that the fax machine report "Commnunication Error". Do you an idea what could be ? Thanks, Angel. > Message: 3 > Date: Sat, 18 Sep 2004 00:48:23 +1200 > From:

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I

Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins dean@collins.net.pr +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download. Thanks Tin Trung Nguyen Technical Team Mobile: 084-91.365.4857 website: www.daivietcontrol.net -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I'm having all sorts of problems with my PHP AGI scripts... Basically, my scripts run fine from the command line and don't do anything well called from Asterisk. Here are my questions: a) Does Asterisk require PHP CLI or CGI? From the command line, my script seems to work fine with PHP 4.3.11 (cli) but not with PHP 4.3.9 (cgi) b) How to debug my script? According

Hi * users, I am only seeing SendText in the available asterisk applications. But I have not seen Receive Text application. I tried on asterisk-1.0.7 and 1.0.9. Can anyone tell me how to use this receive text command. I want to use receivetext command and get text information from an softphone so that that can be routed to some other phone supporting text message. (my soft phones are SIP/IAX

Is it wise to use an outage to promote your business, not on the user's list and not multiple times? Put it in your signature or something ;-) Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Shane Breen > Sent: Wednesday,

There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have 2 outgoing calling companies. Say A and B. With A company, I'm given 250 Minutes free per month. I would like to start off by using A company until 250 minutes were depleted for that month then use B and of course switch back to A once the next month turns up. How would I do this? - -- ================================= Joshua

Guys, is there any way to detect voice when calling a zap channel? For example, if you want to send out or playback a recorded message, you need to wait for somebody to actually answer the phone before playing starts.. Anyway to detect this?

Hello - sorry for my bad english. I will make a list of all sound files on asterisk and i'll record then on professional studio. the french prompts from sineapps sounds bad... sorry for her... tell me if their is many peoples want it ! thank's. en francais: dites moi si ca vaut le coup que j'investissexr dans l'enregistrement des messages en francais. La voix sera la "voix

hello list , my set up is like this ip device -->ser ---> asterisk(astcc) --> pstn gatewsy my asterisk version is 1.0.2 iam using the ser as registration and asterisk aa the prepaid one with the help of the astcc. now my problem is the destination people i.e the pstn line s are listening low voice and also the blurr sound quality along with the audio of the ip device at

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

Hi, Problem: When talking to someone (from pstn) and this person is not talking loud, the voice is cut by Asterisk. I tried increase RxGain but it changed nothing (was talking louder but voice still cut.) I use XLite as soft phone. I think this is probably a codec setting... but how do I check that on server side? I just don't know what to do. All works fine (asteriskathome) but I always

I've been having random problems when I make long distance calls using either VoicePulse or Nufone. Sometimes the calls go through clear, and other calls (or even just part of a call) the person on the other end just hears garbled voice, or really broken up voice. Sometimes it lasts for only a few seconds, but other times it goes on for a few minutes until I give up on the call. At