Nasir Javaid
2010-May-11 08:57 UTC
[asterisk-users] asterisk-users Digest, Vol 70, Issue 23
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
?
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>
Subject: Re: [asterisk-users] Dialing a SIP Peer without using
register strin
To: asterisk-users at lists.digium.com
Message-ID: <hsarab$ok7$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 <abc%3Amysecret at
192.168.0.20><mailto:
abc%3Amysecret at 192.168.0.20 <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
>
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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 > > >