<div>Thanks Vardan,</div>
<div> </div>
<div>I will like to know if this scenario can work when peer is not having fixed ip and we use </div>
<div>host = <a href="http://nasir.server.com">nasir.server.com</a></div>
<div>?</div>
<div>also I have set insecure=invite,port</div>
<div> </div>
<div>what if i use</div>
<div>insecure=no</div>
<div> </div>
<div>thanks again.</div>
<div> </div>
<div>Message: 24<br>Date: Tue, 11 May 2010 10:52:14 +0500<br>From: Vardan <<a href="mailto:hvardan71@gmail.com">hvardan71@gmail.com</a>><br>Subject: Re: [asterisk-users] Dialing a SIP Peer without using<br> register strin<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>Message-ID: <hsarab$ok7$<a href="mailto:1@dough.gmane.org">1@dough.gmane.org</a>><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>Remove username and secret and use IP authentication on both side<br><br>[server1_abc]<br>type=peer<br>host=192.168.0.20<br>context=default<br>dtmfmode=rfc2833<br>canreinvite=yes - canreinvite with nat=yes is not working<br>
insecure=invite,port<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>allow=g729<br>allow=gsm<br>nat=yes<br>qualify=yes<br><br><br><br>[server2_abc]<br>type=peer<br>host=192.168.0.21<br>context=default<br>dtmfmode=rfc2833<br>
canreinvite=yes<br>insecure=invite,port<br>disallow=all<br>allow=ulaw<br>allow=alaw<br>allow=g729<br>allow=gsm<br>nat=yes<br>qualify=yes<br><br><br><br>Nasir Javaid wrote:<br>> Hi,<br>><br>> I am new to this list and this is first time i m posting here. please<br>
> help me out<br>><br>> currently I am working on dialing a sip peer on an asterisk server from<br>> 2nd asterisk server. scenario is like this.<br>><br>> on my system i am using this peer in sip.conf.<br>
><br>> [abc]<br>> type=peer<br>> username=abc<br>> secret=mysecret<br>> host=192.168.0.20<br>> context=default<br>> dtmfmode=rfc2833<br>> ;restrictcid=no<br>> canreinvite=yes<br>> insecure=invite,port<br>
> disallow=all<br>> allow=ulaw<br>> allow=alaw<br>> allow=g729<br>> allow=gsm<br>> nat=yes<br>> qualify=yes<br>><br>> and using following register string<br>><br>> register => <a href="mailto:abc%3Amysecret@192.168.0.20">abc:mysecret@192.168.0.20</a> <mailto:<a href="mailto:abc%253Amysecret@192.168.0.20">abc%3Amysecret@192.168.0.20</a>><br>
><br>><br>> now problem is that when i use register string everything goes ok. but<br>> when i remove register string call doesn't go as expected.<br>><br>> I would like to know if there is any feature that i can use to call sip<br>
> peer and authenticate is in dial command or some feature in sip.conf<br>><br>> i dont wanna use register string. please help.<br>><br>> regards,<br>><br>> Nasir Javaid<br>><br></div>