Göran Törnqvist
2004-Dec-14 04:55 UTC
[Asterisk-Users] How to debug? - SIP calls not coming thru
Hello, I?ve just set up SIP with asterisk using this how-to: http://www.automated.it/guidetoasterisk.htm#_Toc49248757 but when I try calling the number at my SIP provider (Wx3) it doesn?t come thru. I THINK I registered to my SIP provider without any problem, in sip.conf I do: register => My.name:passwd:user@wx3.se/12345678 If I change the password to something else I get an registration error when doing asterisk / reload so I guess the registration went ok the first time then when there?s no error message? I also added 2 phones to asterisk: Cisco IP-phone 7960. I GUESS it was registered successfully because I got errors at first because username was wrong and when I changed it to correct values ? no errors. ?sip show peers? is showing my phones: Name/username Host Dyn Nat ACL Mask Port Status cisco2/cisco2 213.1.1.1 D 255.255.255.255 5060 Unmonitored cisco1/cisco1 213.1.1.1 D 255.255.255.255 5060 Unmonitored wx3.se 213.1.1.1 255.255.255.255 5060 Unmonitored 3 sip peers loaded [3 online , 0 offline] Though host for wx3.se is showing the wrong IP above. How can I debug this? Below is what I?ve added to my config-files. Sip.conf [general] context=mycontext register => My.name:passwd:user@wx3.se/XXXXXXXXX [wx3.se] type=peer fromuser=username-here secret=pass fromdomain=wx3.se [cisco1] type=friend host=dynamic defaultip=192.111.111.111 username=cisco1 secret=mypass dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;mailbox=1000 ; Mailbox for message waiting indicator context=mycontext callerid="MyUser 1" <9053> [cisco2] (almost identical) in extensions.conf: [mycontext] exten => 1,1,Dial(SIP/cisco1,20,tr) exten => 2,1,Dial(SIP/cisco2,20,tr) exten => 12345678,1,Dial(SIP/cisco1&SIP/cisco2,20,tr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041216/e6a4aa69/attachment.htm
Göran Törnqvist
2004-Dec-14 07:07 UTC
[Asterisk-Users] How to debug? - SIP calls not coming thru
Hello, I?ve just set up SIP with asterisk using this how-to: http://www.automated.it/guidetoasterisk.htm#_Toc49248757 but when I try calling the number at my SIP provider (Wx3) it doesn?t come thru. I THINK I registered to my SIP provider without any problem, in sip.conf I do: register => My.name:passwd:user@wx3.se/12345678 If I change the password to something else I get an registration error when doing asterisk / reload so I guess the registration went ok the first time then when there?s no error message? I also added 2 phones to asterisk: Cisco IP-phone 7960. I GUESS it was registered successfully because I got errors at first because username was wrong and when I changed it to correct values ? no errors. ?sip show peers? is showing my phones: Name/username Host Dyn Nat ACL Mask Port Status cisco2/cisco2 213.1.1.1 D 255.255.255.255 5060 Unmonitored cisco1/cisco1 213.1.1.1 D 255.255.255.255 5060 Unmonitored wx3.se 213.1.1.1 255.255.255.255 5060 Unmonitored 3 sip peers loaded [3 online , 0 offline] ?sip show registry? shows: Host Username Refresh State wx3.se:5060 MyUser 105 Registered How can I debug this problem? Below is what I?ve added to my config-files. Sip.conf [general] context=mycontext register => My.name:passwd:user@wx3.se/XXXXXXXXX [wx3.se] context=mycontext type=peer fromuser=username-here secret=pass fromdomain=wx3.se [cisco1] type=friend host=dynamic defaultip=192.111.111.111 username=cisco1 secret=mypass dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info ;mailbox=1000 ; Mailbox for message waiting indicator context=mycontext callerid="MyUser 1" <9053> [cisco2] (almost identical) in extensions.conf: [mycontext] exten => 1,1,Dial(SIP/cisco1,20,tr) exten => 2,1,Dial(SIP/cisco2,20,tr) exten => 12345678,1,Dial(SIP/cisco1&SIP/cisco2,20,tr -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041214/7bf1305e/attachment.htm