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 <hvarda...@gmail.com> Subject: Re: [asterisk-users] Dialing a SIP Peer without using register strin To: asterisk-users@lists.digium.com Message-ID: <hsarab$ok...@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:mysec...@192.168.0.20 <abc%3amysec...@192.168.0.20><mailto: abc%3amysec...@192.168.0.20 <abc%253amysec...@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 >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users