[Asterisk-Users] H323 - SIP Interoperability

[siva kumar]

I DON'T KNOW


>From: "Girish Gopinath" <gopinath_girish at hotmail.com>
>Reply-To: asterisk-users at lists.digium.com
>To: asterisk-users at lists.digium.com
>Subject: RE: [Asterisk-Users] H323 - SIP Interoperability
>Date: Thu, 01 Apr 2004 22:46:10 +0530
>
>Hello,
>
>>From: pesb <pesb at conexion.com.py>
>>Subject: [Asterisk-Users] H323 - SIP Interoperability
>>Date: Thu, 1 Apr 2004 12:37:17 -0300
>
><snip>
>>So, I would like to call SIP/4 phone by dialing 014. Something like this:
>>
>>exten => 01X,1,Dial(SIP/X) ; This is not working
>>
>>How can I do that?
>
>Try this:
>exten => _01X,1,Dial(SIP/${EXTEN:2})
>That should do it.
>
>>Another question: How can I make the RTP data flow go directly from one IP
>>phone to the other? Rigth now, all the RTP data flow goes through the SIP
>>proxy.
>
>set canreinvite=yes for sip users in sip.conf
>
>Regards, Girish
>
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