On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov <[email protected]> wrote: > Hello > > We have recently upgraded to Realtime engine (sip buddies and > extensions) and now have problems with calling local SIP users. > I have rtcachefriends=yes but tried with 'no' and it's even worse. > (asterisk 1.8.1.1 + realtime mysql) > > Here's an example: > > User 1000 registers successfully and can then be called with > Dial(SIP/1000,30) successfully > > After some time when I try to call this user the asterisk just keeps > hanging until timeout occurs: > > -- Calling 1000 > > and the debug says: > > [2010-12-24 12:30:11] DEBUG[12870] chan_sip.c: ** SIP timers: > Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id > #1213)) > [2010-12-24 12:30:11] DEBUG[12870] chan_sip.c: Trying to put 'INVITE > sip:' onto UDP socket destined for 78.84.202.65:48406 > [2010-12-24 12:30:15] DEBUG[12870] chan_sip.c: ** SIP timers: > Rescheduling retransmission 5 to 8000 ms (t1 500 ms (Retrans id > #1213)) > [2010-12-24 12:30:15] DEBUG[12870] chan_sip.c: Trying to put 'INVITE > sip:' onto UDP socket destined for 78.84.202.65:48406 > > however if i do 'sip show peers' it shows the peer normally: > > 1000/nlcyhguv 78.84.202.65 D N 34817 Unmonitored Cached RT > > > User 1000 has nat=yes and is behind NAT. > Before we moved to Realtime it all used to work well. > > > Any advice would be appreciated. > > Thanks in advance, > Nick > > -- > _____________________________________________________________________ > -- 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 >
This is really odd, could you do use a favor and show the full output of "sip show peer 1000 load"? I want to see it to better help you. -- _____________________________________________________________________ -- 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
