Re: [asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer
Can you try a sip.conf entry with the port= parameter as well. For example: [lucent] type=friend host=ip of lucent box port=5060 insecure=port,invite context=default Your INVITE header is including the port, and maybe Asterisk is having trouble matching the sip.conf entry. So the insecure=port,invite option should also include an insecure=user option to disregard any user info in the invite. Is there is another mechanism in Asterisk to disregard any user info from an invite? Thanks. JR -- Andres Technical Support http://www.telesip.net ___ --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
[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue
what is the sip.conf for 1239 which I'm going to assume is a extension on the TNT Barry JR Richardson wrote: Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip sip TNT pri pri asterisk exten 1239 is the CID Number from the originating caller on the PRI side, has no relation to the local user on the sip side of the call. -- JR Richardson Engineering for the Masses ___ --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
[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue
When all else fails I resort to adding this in the sip.conf peer config: Insecure=invite,port It took me a while to figure out they can be used together. Regards, Scott Thanks Scott, i have it set to that, but that has no effect. The incoming call still requires proxy authentication. I've also tried in both general and max context insecure=invite,port autocreatepeer=yes allowguest=yes allowexternalinvites=yes trustrpid = yes There is something different in asterisk 1.0.10 and 1.2. I've tried variations of all sip.conf switches to accept unauthenticated calls and nothing seems to work. I'm wondering is there is a patch that will allow unauthenticated calls in sip? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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
[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer
After mocking up an unauthenticated call from a different device, a spa942 phone, I found something strange in the SIP debug between the phone and the TNT. Asterisk is accepting unauthenticated calls as long as there is not a user in the SIP header from the calling device. Invite from the MAX: does not get passed to the dial plan -- SIP read from 10.10.14.131:5060: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 To: sip:[EMAIL PROTECTED]:5060;user=phone From: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;tag=1e82fc7f-1fb33c15-830e0a0a Remote-Party-Id: NO CID NAME sip:[EMAIL PROTECTED]:5060;user=phone;screen=no;id-type=subscriber;party=calling;privacy=off Call-ID: [EMAIL PROTECTED] CSeq: 803597 INVITE Via: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK007aa1ced4ace55a Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;user=phone Supported: replaces Content-Type: application/sdp Accept: application/sdp Accept-Encoding: Accept-Language: en User-Agent: Lucent-Universal-Gateway Content-Length: 232 Invite from the phone: gets passed to the dial plan in the [general] context= -- SIP read from 10.10.11.51:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.10.11.51:5060;branch=z9hG4bK-9fd7c0a9 From: 2001 sip:[EMAIL PROTECTED];tag=59f6242028d88691o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: 2001 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/SPA942-4.1.12(a) Content-Length: 391 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp The invite string from the TNT: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 The invite string from the phone: INVITE sip:[EMAIL PROTECTED] SIP/2.0 It appears that if a user= field is in the invite message, Asterisk looks for a user context and requires authentication. So the insecure=port,invite option should also include an insecure=user option to disregard any user info in the invite. Is there is another mechanism in Asterisk to disregard any user info from an invite? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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
