[Asterisk-Users] Call recording/SIP not loggin IN

[Chandra]

My sip.conf
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0
disallow=all                    ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference

dtmfmode=rfc2833

[xlite1]
type=user
host=dynamic
secret=xlite1
context=outgoing
reinvite=no
canreinvite=no
qualify=60

[xlite1]
type=peer
host=dynamic
secret=xlite1
reinvite=no
canreinvite=no
qualify=60

In xlite i have User=xlite1, Pwd=xlite1 and SIP Proxy=IP of my * box, Out
bound Proxy= IP of my * box

netstat -na gives

[root at localhost root]# netstat -na
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address           Foreign Address         State
tcp        0      0 0.0.0.0:32768           0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:22305           0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:22273           0.0.0.0:*               LISTEN
tcp        0      0 127.0.0.1:32769         0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:3306            0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:111             0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:2000            0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:5680            0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:22321           0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:22289           0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:21              0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:23              0.0.0.0:*               LISTEN
tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN
tcp        0    128 202.51.xx.xx1:22        202.51.xx.xx0:3148
ESTABLISHED
udp        0      0 0.0.0.0:32769           0.0.0.0:*
udp        0      0 0.0.0.0:5036            0.0.0.0:*
udp        0      0 0.0.0.0:5060            0.0.0.0:*
udp        0      0 0.0.0.0:4569            0.0.0.0:*
udp        0      0 0.0.0.0:111             0.0.0.0:*
udp        0      0 0.0.0.0:11770           0.0.0.0:*
udp        0      0 0.0.0.0:11771           0.0.0.0:*
udp        0      0 0.0.0.0:2427            0.0.0.0:*
Active UNIX domain sockets (servers and established)
Proto RefCnt Flags       Type       State         I-Node Path
unix  2      [ ACC ]     STREAM     LISTENING     1504   /dev/gpmctl
unix  2      [ ACC ]     STREAM     LISTENING     1775
/tmp/.font-unix/fs7100
unix  2      [ ACC ]     STREAM     LISTENING     1520
/var/lib/mysql/mysql.sock
unix  2      [ ACC ]     STREAM     LISTENING     1885
/var/run/asterisk.ctl
unix  2      [ ACC ]     STREAM     LISTENING     1621
/tmp/.iroha_unix/IROHA
unix  2      [ ACC ]     STREAM     LISTENING     1593   /tmp/cd_sockV4
unix  2      [ ACC ]     STREAM     LISTENING     1671   /tmp/kd_sockV4
unix  2      [ ACC ]     STREAM     LISTENING     1699   /tmp/td_sockV4
unix  2      [ ACC ]     STREAM     LISTENING     1565   /tmp/jd_sockV4
unix  7      [ ]         DGRAM                    1094   /dev/log
unix  3      [ ]         STREAM     CONNECTED     1889
/var/lib/mysql/mysql.sock
unix  3      [ ]         STREAM     CONNECTED     1888
unix  2      [ ]         DGRAM                    1778
unix  2      [ ]         DGRAM                    1645
unix  2      [ ]         DGRAM                    1406
unix  2      [ ]         DGRAM                    1160
unix  2      [ ]         DGRAM                    1110
[root at localhost root]#


my grandstream is also not registering to *.

----- Original Message -----
From: "CW_ASN" <cw_asn at fibertel.com.ar>
To: <asterisk-users at lists.digium.com>
Sent: Friday, January 02, 2004 9:14 PM
Subject: Re: [Asterisk-Users] Call recording


> ----- Original Message -----
> From: "Chandra" <chandra at digital.com.np>
> To: <asterisk-users at lists.digium.com>
> Sent: Friday, January 02, 2004 9:30 AM
> Subject: Re: [Asterisk-Users] Call recording
>
>
> > xlite saying login timed out. contact network admin.
> >
> > how to get rid of this. * is not behind NAT.
> >
> > DIAX works fine
> >
>
> Could you especify a bit more?
> Send sip.conf, 'netstat -na' from you linux box, xlite config, etc...
>
> Regards,
>
> Gus
>
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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>
>