[asterisk-users] asterisk-users Digest, Vol 70, Issue 23

[Vardan]

Hello
Yes, you can just remove insecure line, if with out this line is worked
by default insecury=no, so if you not write this line, it will be NO


Also you can use hostname in host field:

===============================================================================
host  = dynamic|hostname|IPAddr
	How to find the client - IP # or host name. If you want the phone to 
register itself, use the keyword dynamic  instead of Host IP.
===============================================================================

like this: host=nasir.server.com

no write  <http://nasir.server.com> in host field.


Vardan



Nasir Javaid wrote:
> Thanks Vardan,
> I will like to know if this scenario can work when peer is not having
> fixed ip and we use
> host = nasir.server.com <http://nasir.server.com>
> ?
> also I have set insecure=invite,port
> what if i use
> insecure=no
> thanks again.
> Message: 24
> Date: Tue, 11 May 2010 10:52:14 +0500
> From: Vardan <hvardan71 at gmail.com <mailto:hvardan71 at gmail.com>>
> Subject: Re: [asterisk-users] Dialing a SIP Peer without using
>         register strin
> To: asterisk-users at lists.digium.com <mailto:asterisk-users at lists.digium.com>
> Message-ID: <hsarab$ok7$1 at dough.gmane.org <mailto:1 at dough.gmane.org>>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> 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> <mailto:abc%3Amysecret at 192.168.0.20
> <mailto:abc%253Amysecret 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
>  >
>