[asterisk-users] asterisk realtime & calling sip users

[Sherwood McGowan]

On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov <nickustinov at 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
>
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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.