actually its a one-way audio issue due to NAT ! alok , please explain your network flow for end to end client-server-client.
You may need to set nat=yes for your sip peer behind NAT. If the server is behind NAT router/firewall use externip=<public.ip.of.server> field. Also provide sip traces of this call. Another thing to do for your learning. Execute wireshark on both softphone systems and set "sip | rtp" as filter and see where are the RTP streams going on each end ! Take a complete capture on Asterisk server by executing the command "sip set debug on" and make a call. BR Sammy On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon <[email protected]>wrote: > alok srivastava wrote: > >> dear >> i have configured properly asterisk. At the one end i am using x-lite >> soft ph and another end twinkle. call is going properly from both end but >> after picking the phone not able to listen other one. >> when i checked the port 5060 on the asterisk server it is always showing >> closed while i have flushed all the rules from iptables (iptables -F) >> >> PORT STATE SERVICE VERSION >> 5060/tcp closed sip >> >> telnet localhost 5060 (could not connect) >> >> regards >> alok >> >> >> SIP is only used to setup (and stop etc.) the call. The actual audio is > sent via rtp. > > /etc/asterisk/rtp.conf > > Should tell which ports asterisk is using for rtp, you will need to make > sure that the remote host can connect to these ports. > > There are lots of articles around on how to resolve this. > > > > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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