nr k
2005-Nov-14 07:53 UTC
[Asterisk-Users] Maximum Number of SIP Phones Supported By Asterisk
Hi All Can anybody tell me the maximum number of SIP Phones supported by Asterisk. regards ramakrishnan.n __________________________________ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs
trixter aka Bret McDanel
2005-Nov-14 08:15 UTC
[Asterisk-Users] Maximum Number of SIP Phones Supported By Asterisk
On Mon, 2005-11-14 at 06:53 -0800, nr k wrote:> Hi All > > Can anybody tell me the maximum number of SIP Phones > supported by Asterisk.If I run asterisk on my ipaq not very many. If I run it on a real server many many more. Your question cant really be answered with the information you have provided. It can only be answered in context. What hardware? What codecs? Any translations (from one medium/codec to another)? What applications are used (AGI, conferences, voicemail, etc)? Is the asterisk server actually pushing the bits for a call or just doing call setup and connecting the two endpoints directly? These are the very minimum questions you have to answer before your question can be answered. There are a few other things that can go into it, but those will help you better define for a rough idea ... And based on the answers to those questions there may be more questions. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051114/dd9b36f1/attachment.pgp
Asterisk guy
2005-Nov-14 12:29 UTC
[Asterisk-Users] Maximum Number of SIP Phones Supported By Asterisk
> > Is the asterisk server actually pushing the bits for a call or just > doing call setup and connecting the two endpoints directly?how to force asterisk just doing call setup and connecting the two endpoints directly? reinvite=yes ? if UA is behind NAT with reinvite=yes , will asterisk actually push the bits or drop the call?