[Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to *

[Mohammed Firdosh Nasim]

On Thu, 2005-03-17 at 11:34, Alexander Lopez wrote:
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mohammed
> Firdosh Nasim
> Sent: Tuesday, March 15, 2005 11:08 PM
> To: asterisk-users at 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
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> 
> 
> 
> 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.