Hi there,
I would like to communicate H323 IP phones with SIP phones. My H323
phones are registered to a gnugk GK, and the SIP phones are registered to a
asterisk SIP proxy.
I could not create a dialplan that works. Inside my extensions.conf file I
created the following two entrances:
exten => 4,1,Dial(SIP/4)
exten => 5,1,Dial(SIP/5)
This allows SIP phones call each other.
And with the following line, the SIP phones can call my h323 phone:
exten => 101,1,Dial(oh323/101)
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?
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.
thanks in advance,
Pablo Salinas
Hello,>From: pesb <pesb@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/${EXTEN:2}) That should do it.>Another question: How can I make the RTP data flow go directly from one IP >phone to the other? Rigth now, all the RTP data flow goes through the SIP >proxy.set canreinvite=yes for sip users in sip.conf Regards, Girish _________________________________________________________________ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home!
I DON'T KNOW>From: "Girish Gopinath" <gopinath_girish@hotmail.com> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: RE: [Asterisk-Users] H323 - SIP Interoperability >Date: Thu, 01 Apr 2004 22:46:10 +0530 > >Hello, > >>From: pesb <pesb@conexion.com.py> >>Subject: [Asterisk-Users] H323 - SIP Interoperability >>Date: Thu, 1 Apr 2004 12:37:17 -0300 > ><snip> >>So, I would like to call SIP/4 phone by dialing 014. Something like this: >> >>exten => 01X,1,Dial(SIP/X) ; This is not working >> >>How can I do that? > >Try this: >exten => _01X,1,Dial(SIP/${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 > >_________________________________________________________________ >Easiest Money Transfer to India. Send Money To 6000 Indian Towns. >http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! > >_______________________________________________ >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_________________________________________________________________ News, views and gossip. Hot downloads ‘n previews. http://www.msn.co.in/Cinema/ Get it all at MSN Cinema!