Francois
2006-Feb-06 11:44 UTC
[Asterisk-Users] Will not authenticate incoming VOIP provider calls
I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I should put insecure=very or insecure=port,invite (depending on the doc). I tried that and a lot of other things, nothing works. That message keeps coming back on every incoming calls. Here are my config files (please don't flame me if there something superobvious I missed, I'm a complete Asterisk newbee). Any help would be greatly appreciated. Thanks SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context=default insecure=very qualify=yes nat=yes host=plasma.digitalvoice.ca register=XXXXXXXXXX:xxxxxx@plasma.digitalvoice.ca/franv [ext1] username=ext1 host=dynamic fromuser = XXXXXXXXXX authname= XXXXXXXXXX fromdomain = plasma.digitalvoice.ca type=friend secret=secret record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never dtmfmode=rfc2833 context=internal canreinvite=no insecure=very callerid=XXXXXXXXXX <xxxxxx> [Digital_out] type=peer secret=xxxxxx username=XXXXXXXXXX host=plasma.digitalvoice.ca fromuser=XXXXXXXXXX fromdomain=plasma.digitalvoice.ca insecure=very context = incoming_calls qualify=yes nat=yes EXTENSIONS.CONF [default] exten => s,1,Answer( ) exten => s,2,Playback(demo-echotest) exten => s,3,Hangup( ) [internal] exten => _NXXNXXXXXX,1,dial(SIP/${EXTEN}@Digital_out,30) exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN}@plasma.digitalvoice.ca,30) exten => 100,1,Playback(demo-echotest) exten => 611,1,Echo( ) [incoming_calls] exten => s,1,Answer( ) exten => s,2,Playback(demo-echotest) exten => s,3,Hangup( )
Colin Anderson
2006-Feb-06 11:48 UTC
[Asterisk-Users] Will not authenticate incoming VOIP provider calls
try host=dynamic in your sip peer entry hth -----Original Message----- From: Francois [mailto:various@desart.ca] Sent: Monday, February 06, 2006 11:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Will not authenticate incoming VOIP provider calls I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129>>From what I've read in the various docs I could access, I should putinsecure=very or insecure=port,invite (depending on the doc). I tried that and a lot of other things, nothing works. That message keeps coming back on every incoming calls. Here are my config files (please don't flame me if there something superobvious I missed, I'm a complete Asterisk newbee). Any help would be greatly appreciated. Thanks SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context=default insecure=very qualify=yes nat=yes host=plasma.digitalvoice.ca register=XXXXXXXXXX:xxxxxx@plasma.digitalvoice.ca/franv [ext1] username=ext1 host=dynamic fromuser = XXXXXXXXXX authname= XXXXXXXXXX fromdomain = plasma.digitalvoice.ca type=friend secret=secret record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never dtmfmode=rfc2833 context=internal canreinvite=no insecure=very callerid=XXXXXXXXXX <xxxxxx> [Digital_out] type=peer secret=xxxxxx username=XXXXXXXXXX host=plasma.digitalvoice.ca fromuser=XXXXXXXXXX fromdomain=plasma.digitalvoice.ca insecure=very context = incoming_calls qualify=yes nat=yes EXTENSIONS.CONF [default] exten => s,1,Answer( ) exten => s,2,Playback(demo-echotest) exten => s,3,Hangup( ) [internal] exten => _NXXNXXXXXX,1,dial(SIP/${EXTEN}@Digital_out,30) exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN}@plasma.digitalvoice.ca,30) exten => 100,1,Playback(demo-echotest) exten => 611,1,Echo( ) [incoming_calls] exten => s,1,Answer( ) exten => s,2,Playback(demo-echotest) exten => s,3,Hangup( ) _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users