I have opened up all the ports specified other than the 10000-20000 range as my router just can't cope with that. Unfortunately I still get no sound.
Is IAX the best route or is registering my FWD connection through SIP.conf the best solution, what do people recommend? I have dug through the WIKI, and what instruction I can find, but to no avail, examples or working confs would be fantastic just for comparison. Thanks Again Chris -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Eastman Sent: 16 August 2004 00:44 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing? IAX2 uses udp port 4569, so you’ll probably have to open that up on your firewall/router. http://www.voip-info.org/ is a good starting place for any asterisk problems - specifically: http://www.voip-info.org/wiki-Asterisk+firewall+rules http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD HTH Ed ________________________________________ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Blunt Sent: 15 August 2004 23:06 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Inbound Free World Dialup - extension not ringing? Hi Lyle, Thank you so much for your help, I think your information points to using IAX2 rather than registering with FWD from the sip.conf I have made an attempt to understand this, added the appropriate information into iax.conf, remove old info from sip.conf, gone to fwd and ticked the IAX registration box, and I now get my local sip phone ringing when I dial in from FWD! Hurrah, unfortunately I get no sound in either direction. Do you have any experience of this or could it be due to me being inside a NAT firewall? I have port 5060 forwarded to my * server, should I forward any other ports? (I can only forward a maximum 20 single ports due to a limitation on my home router). As yet I am unable to make outgoing calls over FWD, I figured I would look at this next. Is there a NAT solution that could be used with sip.conf rather than the IAX? Again your help is most appreciated. Best regards Chris ________________________________________ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: 15 August 2004 15:14 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Inbound Free World Dialup - extension not ringing? You need a defination for the inbound FWD and what to do with that. In my extensions.conf, I have: [globals] FWDNUMBER=123456 ; your actual fwd number FWDCIDNAME='My Name' FWDPASSWORD=myfwdpasswd FWDRINGS=sip/office FWDVMMBOX=1010 [fwd_out] exten => _123.,1,SetCallerId,${FWDCIDNAME} ; replace 123 with the desired access code to dial out via FWD exten => _123.,2,Dail(IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN}:3},60 ,r) exten => _123.,3,Congestion [local] include => fwd_out :add to local context [default] ;inbound dialing from FWD exten => ${FWDNUMBER},1,Goto(housemenu,s,1) ; I have mine set to hit a menu, no reason you cann't forward to an extension instead ----- Original Message ----- From: Chris Blunt To: [EMAIL PROTECTED] Sent: Sunday, August 15, 2004 3:29 AM Subject: [Asterisk-Users] Inbound Free World Dialup - extension not ringing? Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux – But I am learning fast. My config is quite simple, I’m just following examples and the Wiki: I have two PC’s running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to set up Asterisk to accept calls from FWD on another number I have registered, but I can’t get my local X-Lite to ring on an inbound call from FWD, and I get the busy tone on the BT100 When I sip debug, I can see that I am registered with FWD, and when I call the number from the BT100 I can see all the incoming information but still nothing on my X-Lite. My extensions.conf: [general] static=yes writeprotect=no [globals] [sip] exten => 1,1,Dial(SIP/phone1,20,tr) exten => 2,1,Dial(SIP/phone2,20,tr) exten => 2,2,VoiceMail,u1234 exten => 2,102,VoiceMail,b1234 ;exten => 1000,1,Dial(SIP/phone1&SIP/phone2,20,tr) exten => 1001,1,Ringing exten => 1001,2,Wait(2) exten => 1001,3,VoicemailMain,s1234 exten => 6601,1,WaitMusicOnHold(60) exten => 232999,1,Dial(SIP/phone1,30,tr) exten => 232999,2,Hangup I am behind a NATed fire wall, but I’m not sure that is related. Any ideas or help (working simple confs) would be much appreciated. Best regards -- Chris Blunt SIP: [EMAIL PROTECTED] _______________________________________________ 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 _______________________________________________ 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
