Hi, I hope you might give me some hints on how to find where my configuration is wrong, I am new to Asterisk and do not know, how to find the problem.
Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the same maschine: Hylafax fax server. I want hylafax to use t38modem (a virtual T.38 modem) for sending faxes. t38modem schould connect to asterisk on the same host. If hylafax sends a fax it should use t38modem which ist connected to asterisk. Asterik is expected to establish an outbound connection to my SIP provider which supports T38. The asterisk box is behind nat. For some reason, t38modem tells hylafax the line is BUSY so there is no fax send. I don't know why there is a busy signal, maybe the call forwarding configuration is wrong, maybe the registration on my SIP provider fails, maybe ....? I simply don't know how to debug what's going on. If Asterix trying to establish an outgoing connection ... Maybe you can help to enlighten me :-) ---------[ sip.conf ]---------------------------------------------- [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes localnet=192.168.0.0/255.255.255.0 localnet=127.0.0.1 externhost=hostname.no-ip.org ;disallow=all ;allow=ulaw ;allow=alaw language=de nat=yes ; incoming register => 4953610000000:[email protected]/4953610000000 ; local SIP-Account where t30modem registers [30] callerid=T38modem<30> host=dynamic ;domain=127.0.0.1 ;host= 127.0.0.1 :permit=127.0.0.1 user=30 secret=password type=friend ;mailbox=30 nat=no context=fax_out ;port=6060 canreinvite=no t38pt_udptl=yes [20] ; FritzBox ; this is an ATA, but this entry is ; probably not needed; the ATA does not register ; a SIP account on asterisk. callerid=FritzBox<20> type=friend username=20 secret=password host=192.168.0.222 fromuser=20 canreinvite=no qualify=no disallow=all allow=alaw allow=ulaw ;allow=ilbc allow=g726 ;allow=g729 allow=gsm ;insecure=very nat=no dtmfmode=info ;tos=0x18 ; Outgoing calls to my SIP provider [4953610000000] type=friend username=4953610000000 secret=password host=sip.1und1.de fromuser=4953610000000 canreinvite=no qualify=no disallow=all allow=alaw allow=ulaw ;allow=ilbc allow=g726 ;allow=g729 allow=gsm ;insecure=very nat=yes dtmfmode=info tos=0x18 ---------[ end of sip.conf ]--------------------------------------- ---------[ extensions.conf ]--------------------------------------- [general] static=yes writeprotect=no [1und1-fax-out] exten => _0.,1,Dial,SIP/${EXTEN}@4953610000000|45|r [default] include => 1und1-fax-out ---------[ end of extensions.conf ]-------------------------------- Any idea what might be wrong? Thank you very much! Sebastian -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
