I am still having this problem of only capturing part of the IP address, I am currently checking into a possible hardware/software issue on the client side but was wondering if there are any setting I need to set on the asterisk server to allow an peer to peer call. I have set dtmfmode=inband. Is there anything else I need to set?
Brian > Message: 5 > From: "David J Carter" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Making a SIP call > Date: Sat, 22 May 2004 00:14:55 +0100 > Reply-To: [EMAIL PROTECTED] > > Check your sip.conf > > Make sure the dtmfmode is set the same as the phone. > > I had this before. > > Usually to dial an IP address you have a keystroke before you enter the > address. > I think on a Grandstream phone you press the menu button then the IP > address. > > Dave > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of > [EMAIL PROTECTED] > Sent: 21 May 2004 21:57 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Making a SIP call > > > If someone could point me in the right direction I would much appreciate > it. Here is my problem: > > My directions for my sip phone says to dial an ip address 12*34*65*78#. > When I dial that into my phone my asterisk server is only picking up some > of the numbers in the above example it would pick up 6578. Then of course > not find it and ring busy on the phone. The same is true for dialing a > regular phone number ( it seems to pick up 4 digits or so) > > I very new to setting this up so I imagine I need to make a change to a > config file, but don't know where to start. _______________________________________________ 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
