[asterisk-users] asterisk realtime & calling sip users

[Nick Ustinov]

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.mcgowan at gmail.com> wrote:
> 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.
>
> --
> _____________________________________________________________________
> -- 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
>
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