[asterisk-users] Dialing a SIP Peer without using register strin

[Vardan]

Remove username and secret and use IP authentication on both side

[server1_abc]
type=peer
host=192.168.0.20
context=default
dtmfmode=rfc2833
canreinvite=yes - canreinvite with nat=yes is not working
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes



[server2_abc]
type=peer
host=192.168.0.21
context=default
dtmfmode=rfc2833
canreinvite=yes
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes



Nasir Javaid wrote:
> Hi,
>
> I am new to this list and this is first time i m posting here. please
> help me out
>
> currently I am working on dialing a sip peer on an asterisk server from
> 2nd asterisk server. scenario is like this.
>
> on my system i am using this peer in sip.conf.
>
> [abc]
> type=peer
> username=abc
> secret=mysecret
> host=192.168.0.20
> context=default
> dtmfmode=rfc2833
> ;restrictcid=no
> canreinvite=yes
> insecure=invite,port
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> nat=yes
> qualify=yes
>
> and using following register string
>
> register  => abc:mysecret at 192.168.0.20 <mailto:abc%3Amysecret at 192.168.0.20>
>
>
> now problem is that when i use register string everything goes ok. but
> when i remove register string call doesn't go as expected.
>
> I would like to know if there is any feature that i can use to call sip
> peer and authenticate is in dial command or some feature in sip.conf
>
> i dont wanna use register string. please help.
>
> regards,
>
> Nasir Javaid
>