On Thu, 2005-03-17 at 11:34, Alexander Lopez wrote: > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Mohammed > Firdosh Nasim > Sent: Tuesday, March 15, 2005 11:08 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g. > WindowsMessenger) from different subnet to * > > On Sat, 2005-03-12 at 07:42, Luki wrote: > > Firdosh, > > > > there were couple typos on my last email, but that's essentially what > > I said. There are two ways of doing it -- but neither will work given > > you current setup. > > > > 1) Phone A talks directly to B. > > 2) Both Phone A and B talk to a common point C. Point C proxies > > traffic between A and B, because A and B cannot see each other > > directly. > > > > You you can't have both clients on the same subnet, then you need a > > third subnet C that is reachable from both A and B. Asterisk runs in > > subnet C and proxies the traffic between A and B. > > > > --Luki > > > Hi All, > > I have a dedicated * server at 172.16.200.150 and my two windows > messenger clients are at 172.16.25.X & 172.16.15.X. Now the server is > visible to both the subnets.Both the users/clients[say msn1 & msn2] are > configured. Then call is made from one user to another. After the callee > receives/accepts the call, neither of users able to hear anything. Sip > debug shows 200 OK for the call.Do I have to "register=>" the users, if > yes kindly mail the register string. > > Here are the sip.conf and extensions.conf > > sip.conf > --------- > [msn1] > type=friend > host=dynamic > context=default > dtmfmode=inband > disallow=all > allow=ulaw > allow=alaw > canreinvite=yes > nat=yes > > > > > [msn2] > host=dynamic > type=friend > context=default > dtmfmode=inband > disallow=all > allow=ulaw > allow=alaw > canreinvite=yes > > extensions.conf > ---------------- > [default] > exten => msn1, 1, Dial(SIP/msn1, 20) > exten => msn2, 1, Dial(SIP/msn2, 20) > > > > Thanks and regards, > > Firdosh > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > For starters, Get rib of canreinvite=yes, set it to canreinvite=no. This > will keep * in the Media path. (You can try msn1 to msn2 directly later) > > Second, what does the output of 'sip show peers' show?? This is from the > CLI prompt on the asterisk server console.
> I just changed canreinvite=yes to canreinvite=no and its working fine. Thanks a lot for ur suggestion. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users