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Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

Hi, Please take a look at the following extensions.conf:- exten => _11XXXX,1,NoCDR() exten => _11XXXX,2,Dial(SIP/${EXTEN},10) exten => _11XXXX,3,VoiceMail() I'm already using realtime for some extensions/users/voicemail. Is there any way to do the following at point 3?:- Lookup the realtime users db and read the MailBox column for that buddy. If the mailbox column is empty, play

Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one? Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/3bee2776/attachment.htm

Does anyone know where i can find out how powerful a machine has to be to handle a certain amount of call volume? Eg, 2Ghz is enough processing power to maintain 100 calls at a time. 4Ghz is engouh to process 250 calls etc etc. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys, Basically, i'd like to be able to force a call to drop by using one of the following methods. Does anyone know if it is possible? It has to be more or less realtime. a) Issuing a command to asterisk via telnet. b) Altering a field in the realtime database c) Any other suggestion or possible method. Thanks in advance. Dan -------------- next part -------------- An HTML attachment

I have the following setup: Asterisk Server Sip Software Phones and a Wholesale connection At the moment, i can receive and send calls through the wholesale connection to and from the SIP phones. Now i want to collect the CallerID from the wholesale connection, and pass it on to the SIP phones when calls are routed through to them. I've tried a number of things including the line below,

I set up my * server using its publc IP address. Now that i switch over to using the domain name, X-Lite can't log in. =========With Domain Name (doesnt work)============ Transmitting (NAT) to 85.250.206.46:6007 <http://85.250.206.46:6007>: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.250.179.93 <http://85.250.179.93>

Hi, Does anyone know how to play music to a caller while you dial a second call? Once the second calls has answered, i'd like to music to stop, and the calls to be bridged. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060130/eb3402df/attachment.htm

Following is the last few lines of output when i try to launch Asterisk:- [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_saycountpl.so] => (Say polish counting words) == Registered application 'SayCountPL' [func_cut.so] => (Cut out information from a string) == Registered custom function CUT == Registered custom

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

The following occurs during make asterisk-addons. I'm ok with asterisk but debugging things like this isnt my strong point. Can anyone give me a pointer? Thanks Dan Journo [root@sip1 src]# cd asterisk-addons [root@sip1 asterisk-addons]# make make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes