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Hi! Probably another newbie question. Is it possible to run * on one processor and MySql on the other in a double cpu server? Anders -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/6e3590b5/attachment.htm

It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

Hi all. Is it possible to get * to send calls to different sip trunks depending on what codec the incoming call use? This to avoid transcoding Anders -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050924/0e209878/attachment.htm

Hi! Is there a simple way for an * newbie to force * to use different sip-trunks for different calls. I have 2 siptrunks, one for inland calls and one for international calls. All in country numbers starts with 0 and all international starts with 00. This I have configured in the outbound routing. But * always use the incountry trunk because the 0. dialpattern is also true for international

I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxxxxxxxxxx type=peer username=0406082250 Regards Anders Svensson -------------- next part -------------- An HTML attachment was scrubbed... URL: