after some deep tracing it turned out to be a faulty router problem thanks.
On Sun, Dec 26, 2010 at 9:38 AM, Sherwood McGowan <sherwood.mcgo...@gmail.com> wrote: > On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov <nickusti...@gmail.com> 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 -- _____________________________________________________________________ -- 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