hi all how can i put a sip proxy; how it differs from asterisk pbx; currently doing some test on asterisk; thru DSL connection the callers can hear only one way;asterisk pbx is behind NAT; i am in search of a proper VOIP network ;appreciate some clues in this line thanks -kbh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120303/c0a7b609/attachment.htm>
On Sat, 3 Mar 2012, K bharathan wrote:> how can i put a sip proxy;Depending on your OS and which proxy you want, it may be as simple as 'sudo yum -y install opensips'> how it differs from asterisk pbx;I think the major distinction is 'who handles the signaling and who handles the media.' A SIP proxy usually only handles the signaling. A PBX usually handles both. Note that you can combine a SIP proxy and a RTP proxy. Then the distinction is based more on the feature set with a PBX having more features relating to handsets.> currently doing some test on asterisk; thru DSL connection the callers > can hear only one way;asterisk pbx is behind NAT;A common NAT problem. Using 'wireshark' to capture packets and analyze them may yield some clues. Recent discussions on this list incriminate SIP 'helpers' in some routers.> i am in search of a proper VOIP network ;appreciate some clues in this > lineEvery one of 'them' is different. How big of a system are you designing? Astlinux on a Soekris is great for a couple of simultaneous calls. Multiple OpenSIPS hosts fronting a farm of Asterisk hosts could probably handle a small nation. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
On Sat,Mar 03 09:17:PM, K bharathan wrote:>thru DSL connection the callers can hear only > one way;asterisk pbx is behind NAT;Greetings, If the calls are esteblished,then SIP did its work. The system may RTP problems, NAT may or may not be the casue for the issue. Defaults RTP ports for asterisk are UDP 10000-20000, You may want to look into that direction. Guy Gold