Thanx for the tip. It has erased the problem i was having using authentication but another problem has arised. i am now able to call with only one user from AST1 to AST2. If i dial using the other user, my AST2 shows the following warning and responds with a "403 forbidden" sip response:
*WARNING[13520]: chan_sip.c:8117 check_auth: username mismatch, have <adf>, digest has <abc>* Any solutions to this problem? On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky <[EMAIL PROTECTED]> wrote: > Rizwan Hisham wrote: > > I am having a strange problem. I am using my asterisk server AST1 to > > register with another asterisk server AST2 using 2 accounts (2 register > > commands in sip.conf). I have made 2 local users in AST1, and configured > my > > dialplan in such a way that both local accounts on AST1 use different > trunks > > to send the call to AST2 server. These 2 different trunks are for 2 > accounts > > i have registered on AST1. > > (skiped) > > > > How can i make asterisk realize it? > > > You must enable authentication of INVITE that AST1 send to AST2. Now you > have no authentication of incoming INVITE and AST2 make decision about > used account based only on IP address of caller peer. > > Changing insecure=port,invite to insecure=port should help. > > -- > Best regards, > Igor A. Goncharovsky > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards Rizwan Hisham
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