[asterisk-users] asterisk realtime & calling sip users

[Nick Ustinov]

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