[asterisk-users] IAX vs SIP - Getting Asterisk out of the media path

Hugo Livude hugolivude at gto.net
Fri Feb 16 20:29:16 MST 2007


If a call comes into my Asterisk server on a DiD provided by an ITSP and the
dialplan sends that call to another external number throught the same ITSP's
network, I don't want the RTP packets to pass through my server once the
call is bridged.

I have had great success getting this to work using IAX, but I have not been
able to get this to work with SIP.   The call is bridged OK (media at both
ends) but the media continues passing through my network.

The default behaviour for the Dial command is to have Asterisk step out of
the media path provided you avoid some options like tT, which I do, so this
should work.

One interesting note: In an Ethereal trace, I see "407 Proxy Authentication
required" just after the INVITE to the callee.  Could that be part of the
problem?  If so what's the fix?  I thought it had something to do with the
"auth" parameter.

I am:

- Behind a NAT,
- Running Red Hat 9.0
- Running Asterisk 1.2.14

How do I stop the media passsing through my Asterisk server after a call
between two external parties has been bridged?

My sip.conf and the dial command I use are below.

Thanks,
Hugh

;*********************** Dial Command *******************************
exten => _6136930630,n,Dial(SIP/6137451576 at 6135551234)

;************************ SIP.conf **********************************
[general]
;
context=incoming-bogus-calls
bindport=5060
bindaddr=0.0.0.0
maxexpirey=3600
defaultexpirey=3600
notifymimetype=text/plain
rtptimeout=60
rtpholdtimeout=300
disallow=all
allow=ulaw
;
; This section is because i'm behind nat
;
externip=999.99.999.99 ;Outside address
localnet=192.168.0.148/255.255.255.0 ;Inside Network
;
register=>6135551234:99999999 at sip02.myITSP.ca/6135551234
;
[6135551234]
type=peer
;auth=md5
auth=6135551234:99999999 at sip02.myITSP.ca
username=6135551234
fromuser=6135551234
fromdomain=myITSP.ca
secret=99999999
host=sip02.myITSP.ca
port=5060
nat=yes
canreinvite=yes
qualify=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=very
context=incoming-sip
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.411 / Virus Database: 268.17.36/681 - Release Date: 11/02/2007



More information about the asterisk-users mailing list