[EMAIL PROTECTED] wrote: > Rich Adamson schrieb: >>> In the past I had problems with the audio over sip. Then I tried the >>> "-p" Option and increased the memory. Now it is better but not >>> perfect. >>> >>> Are there any more possibilities to increase it more? By now I'm >>> using a P-II/333. >>> >>> Could a completely hand optimized kernel (I use 2.6.) help a bit? >> >> There's no way to answer your question with any degree of reasonable >> truth as you haven't mentioned they type of phones, type of pstn >> interface, which codecs, etc, etc. > > Okay. My server has got: > > - One Phonejack Lite > - One X100P Clone > - 256mb Memory > - P II/333 > - Linux 2.6.5 > - Debian Woody > - Asterisk 1.0.1 > - Codecs: GSM, ulaw, alaw > - ADSL 1000kBit/s Downstream, 128kBit/s Upstream
That upstream bandwitch will need to be managed carefully. If you're using G.711, one channel would be using roughly 80kbit of your upstream. Who has the most quality complaints: you, or the people you are talking to? > Calls from or to the pstn are completely okay. Calls over SIP aren't. > Calls over IAX couldn't be tested at the moment. Can you make a SIP connection directly to the box? No LAN, no WAN, just a crossover cable between your SIP phone (soft or hard) and your Asterisk system? That'll give us some idea of whether the problem is network or server-based. Jim. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.0 - Release Date: 17/12/2004 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
