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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120701/3907e9eb/attachment.htm>
Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet is using TCP. I am typing from my mobile phone... Il giorno 01/lug/2012 09:35, "alok srivastava" <alokkic at gmail.com> ha scritto:> 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 > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120701/71eb63d9/attachment.htm>
No voice means you have to look at the rtp ports. You can find more via google "firewall rtp ports asterisk" B. Op 1-7-2012 9:34, alok srivastava schreef:> 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 > > > > -- > _____________________________________________________________________ > -- 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 >
On Sun, 2012-07-01 at 13:04 +0530, 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 > alokHi Alok, telnet is a very crude tool to test with. Try hping or nmap instead. Hans
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.
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 <digium at sanguinarius.co.uk>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> >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120702/536657f6/attachment.htm>