I had the same problem. But I had a Sip 400 Bad Result ... Failed to authenticate on INVITE ...
I am running asterisk 1.03 I do not have my program pointing to any proxies...It is pointing to Sip.broadvoice.com, I do not have any proxies set up in my /etc/hosts file where PPPPPPPPPP = Phone Number XXXXXXXXX = secret Try cutting and pasting mine in, see if it works... My Sip.conf is as follows: [general] context=sip ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls dtmfmode=inband disallow=all allow=ulaw allow=gsm register => PPPPPPPPPP:[EMAIL PROTECTED] [PPPPPPPPPP] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PPPPPPPPPP secret=XXXXXXXXXX username=PPPPPPPPPP insecure=very context=sip authname=PPPPPPPPPP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no On Thursday 10 March 2005 02:08 pm, Joe wrote: > Zanzamar, > > I agree that it should work. I can call out and have the land phone > ring, but as soon as it is answered, another invite goes out and that > is when I get the 401 not authorized. I don't want to go down this > route, but could this be a Codec issue? > > Here is my sip config > > > > [sip.broadvoice.com] > type=peer > host=sip.broadvoice.com > fromdomain=sip.broadvoice.com > fromuser= BBBBBBBBBB > username= BBBBBBBBBB > authuser= BBBBBBBBBB > secret= secret > context=sip > nat=no > insecure=very > dtmfmode=inband > > > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
