asterisk users - Mar 2005 - How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to *

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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
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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.

On Thu, 2005-03-17 at 11:34, Alexander Lopez wrote:
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] 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.