[asterisk-users] Sip module problem

Luis Silva luis.silva at dreamware.pt
Tue Mar 2 05:14:32 CST 2010


Hi,

I need some help debugging a sip situation.

I started to have problems with sip trunks, using more than one trunk (and
sometimes using only one) the sip module seems to freeze.

My local extensions lost registration and also the trunks.  The only way
that I can restart the sip is removing the trunks. If I make sip reload or
restart asterisk the sip module takes many many time before starting.

I use the logger in full in order to debug the problem but don't see
anything strange. 

 

During the freeze If I make sip show peers , all the extensions and trunks
are unreachable, (where etx 10 and 11 and trunk telepac5)

 

12/12                      (Unspecified)    D   N      0        UNKNOWN

11/11                      172.16.1.100     D   N      5063     UNREACHABLE

100                        (Unspecified)    D   N      0        UNKNOWN

10/10                      172.16.1.101     D   N      25124    UNREACHABLE

telepac5/+351302028197     213.13.89.67                5060     UNREACHABLE

 

But I know that are request's coming to my box if I check with netstat I'm
receiving packages  

 

[root at localhost ~]# netstat -an|grep 5060

udp        0      0 0.0.0.0:5060                0.0.0.0:*

[root at localhost ~]# netstat -an|grep 5060

udp     1840      0 0.0.0.0:5060                0.0.0.0:*

[root at localhost ~]# netstat -an|grep 5060

udp     1840      0 0.0.0.0:5060                0.0.0.0:*

[root at localhost ~]# netstat -an|grep 5060

udp     1840      0 0.0.0.0:5060                0.0.0.0:*

[root at localhost ~]# netstat -an|grep 5060

udp    27600      0 0.0.0.0:5060                0.0.0.0:*

[root at localhost ~]# netstat -an|grep 5060

udp    33120      0 0.0.0.0:5060                0.0.0.0:*

 

The second field  Recv-Q is according to man "The count of bytes not copied
by the user program connected to this socket.", so this means that asterisk
is not getting this packets right?...

 

 I started with asterisk version 1.4.26.1, upgraded to 1.4.29 but got the
same result

 

How can I debug this situation? Is there a way to debug the "insides" of the
sip module?

 

Regards

Luis

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