Hot answers tagged asterisk
Turns out the fix was fairly simple—a BIOS upgrade. Upgrading from 0801 to 1406 (the most recent as of this writing, from 2009) fixed it. Direct link to the 1406 BIOS update. Or the P5N-E SLI download page should have the latest.
Yes, this is possible with some dialplan scripting. You would want to use the Dial() application to "pick up and dial the 1st number". With a dumb analog line, you might not get any indication that the call was answered by the remote side. You can either have the first callee enter a DTMF digit when they pick up to indicate that it's time to call the next ...
Try using sar (http://www.thegeekstuff.com/2011/03/sar-examples/) to track cpu usage, and also the Asterisk CDR to track call records. If you find a correlation between high CPU usage and dropped calls, then yes you are likely right. I suppose it is possible that the load average feature isn't working right, virtual servers can be a bit funky that way in ...
For polycom phones have be Alert-Info: Ring Answer Or have be in config <voIpProt> <alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1./> You also can change that option via phone's web interface
Acording to this blog post titled: How to install Asterisk 1.8.4 on Debian 6.0.1 I think you want to install ncurses-dev and not libncurses5-dev. They're building an older version of Asterisk but thought this might be of help in your situation. Also you can see all the ncurse packages like so: $ apt-cache search ncurses | grep dev
You can connect asterisk channel to asterisk-alsa(local audio card of server) After that you can connect sound card output to audio-in "Analog L&R In 6 (including Front mini jack input)" at your system.
If you can get line-level audio to it, of course. If you want to send a wav (or mp3) over the network, then you have some digging to do. Click the "download" tab on the Morantz page you linked to find the remote command set to control the receiver. The most direct option would have asterisk run a script to send the necessary commands to the receiver, ...
You can use it in PBX on FXS port, also you can create hot line using such gateway connected to pstn. You just call pstn number and got call somewhere to gsm(operator's phone)
Here's what I would set in sip.conf register => username:email@example.com [myprovidername] host=sip10.provider.com outboundproxy=sip10.provider.com:5090 type=friend fromuser=username defaultuser=username secret=password context=myproviderinbound Regarding "register =>" I don't know if your sip provider requires it, but... My guess is that ...
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