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If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

Hi, I have the newbie guide from AMP's website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ? Any help appreciated. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/6b7a2f61/attachment.htm

Hi All, I am rather new to the asterisk world, and new to VoIP in general, my question seems rather simple compared to some of the topics under discussion here :) I have done quite a bit of reading and fiddling trying to get a system set up, to no avail yet basically myself and a friend (both behind NAT and Firewalls, but able to set up the firewall rules/port mappings ourself) are interesting in

>From couple of weeks i am working on asterisk fax but was not successful. I am able to receive only half fax documents and its ending with blurred lines. I have tried with almost all the versions of spandsp 0.0.2pre4,pre10 etc but of no use. I have digium card to which i have plugged in 2 pstn lines and i have removed echocancellation too(zapata.conf) and also enabled g7111 alaw&Ulaw

Hi all, After pinging the AMP userlist at SourceForge, I got a great step by step explanation as to how to set up AMP for Apache2 (some maybe obvious stuff that wasn't in the Newbie Guide). Thanks to Jason Becker of Coalescent Systems. If anyone needs me to post Jason's instructions here, I can, but they are in a thread called "AMP noob issues with Apache2/Suse9.2" at

Hello, Has anyone used any 3rd party web based software to get performance information out of Asterisk? Looking for CPS, call setup times, voicemail database utilization etc. Cheers Keith. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050208/e53730f8/attachment.htm