Configuration issues - can't login with SIP


#1

Hi group

I have quite a problem with my Asterisk (I am new to VoIP).

sip.conf

[general]
context=default
disallow=all
allow=ulaw
allow=alaw
allow=gsm
port=5060
bindaddr=0.0.0.0
register => 46933245:XXXXX@musimi.dk/900

; SIP profile for remote SIP host musimi.dk
[musimi.dk]
type=friend
host=musimi.dk
username=46933245
secret=XXXXXX
fromuser=46933245
fromdomain=musimi.dk
context=musimi_incoming
nat=yesqualify=10000
canreinvite=no
insecure=very
dtmfmode=rfc2833

; SIP profile for local X-Lite
[xlite]
type=friend
host=dynamic
context=dialout
username=xlite
secret=xlite
callerid=“Preben Holm” <2>
nat=no; LAN
qualify=yes

; SIP profile for local KPhone
[101]
type=friend
host=dynamic
context=dialout
username=101
secret=101
callerid=“Preben Holm” <101>
canreinvite=no
nat=no
qualify=yes

extensions.conf

[general]
static=yes
writeprotect=no

[globals]

[dialout]
include => internal
include => musimi_outgoing

[musimi_outgoing]
exten => _0XXXXXX.,1,Dial(Sip/musimi/${EXTEN:1},120)
exten => _0XXXXXX.,2,Congestion

[musimi_incoming]
exten => 900,1,Dial(Sip/102,120)
exten => 900,2,Congestion

[internal]
exten => 101,1,Dial(Sip/101,120)

I have it registering with my VoIP provider, but the configuration locally doesn’t seem to be working.

The output from asterisk —vvgc is:

== Parsing ‘/etc/asterisk/extensions.conf’: Found
– Registered extension context ‘dialout’
– Including context ‘internal’ in context ‘dialout’
– Including context ‘musimi_outgoing’ in context ‘dialout’
– Registered extension context ‘musimi_outgoing’
– Added extension ‘_0XXXXXX.’ priority 1 to musimi_outgoing
– Added extension ‘_0XXXXXX.’ priority 2 to musimi_outgoing
– Registered extension context ‘musimi_incoming’
– Added extension ‘900’ priority 1 to musimi_incoming
– Added extension ‘900’ priority 2 to musimi_incoming
– Registered extension context ‘internal’
– Added extension ‘101’ priority 1 to internal
Reloading MGCP
== Parsing ‘/etc/asterisk/mgcp.conf’: Not found (No such file or directory)
Unable to load config mgcp.conf, MGCP disabled
Reloading SIP
== Parsing ‘/etc/asterisk/sip.conf’: Found
Found route to 212.130.58.214, output from our address 192.168.1.2.
Stopping retransmission on ‘2d816ca67d9a06e7665982946c5de13d@192.168.1.2’ of Request 102: Found
Setting NAT on RTP to 4
Found route to 212.130.58.214, output from our address 192.168.1.2.
Scheduled a registration timeout # 17
Stopping retransmission on '02762a833196340018cc54396dbf6a7e@musimi.dk’ of Request 102: Found
Stopping retransmission on '02762a833196340018cc54396dbf6a7e@musimi.dk’ of Request 103: Found
Registration successful
Cancelling timeout 17
Found route to 192.168.1.50, output from our address 192.168.1.2.
Registration from ‘“101” sip:101@192.168.1.2’ failed for '192.168.1.50’
Auto destroying call ‘886126356@192.168.1.50’

btw. what does the retransmission thing do, and why is it there?

I try logging in with KPhone!

Btw. the name in the brackets [101] can that be different from the username and the callerid=“Preben Holm” <101>?

Thanks for all you help?