Hi Everybody
Any significant changes to CVS HEAD over the last couple of days. I've got
two asterisk boxes - both on public IP but one is dynamic. The one on
dynamic IP registers at the other one - that part is fine.
Calls going from the one with dynamic to the static one goes fine.
Call the other way results now in:
Failed to authenticate user "1101" <sip:mydk@xxx.xxx.xxx.xxx>
1101 is a SIP phone authenticated at the static server. All sip entries
have "canreinvite=no". Two days ago this was working fine.
Regards,
Lars...
--
Lars Boegild Thomsen
Technical Director
JustIT Sdn. Bhd.
Cell Phone (MY): +60 (16) 323 1999
ICQ: 6478559
Yahoo Chat: lars_boegild_thomsen@yahoo.com
MSN Chat: lars_boegild_thomsen@hotmail.com
http://www.justit.ws
Phone: +1 (360) 515 3551 (US) +45 8692 1951 (DK) +60 (3) 2057 2646 (MY)
Fax : +60 (3) 2057 2647 (MY)
Hi Lars, I met the same problems yesterday and even posted it to the list. Unfortunately nobody answered yet. Is it so clear to solve that no one is willing to help us? :-/ Regards, Julian Pawlowski
Julian Pawlowski [lists@jp-solution.net] wrote:> I met the same problems yesterday and even posted it to the list. > Unfortunately nobody answered yet. > > Is it so clear to solve that no one is willing to help us? :-/ >It sometimes helps if you quote some context above your text. -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ kevin@cursor.biz _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/
Lars Boegild Thomsen wrote:> Hi Everybody > > Any significant changes to CVS HEAD over the last couple of days. I've got > two asterisk boxes - both on public IP but one is dynamic. The one on > dynamic IP registers at the other one - that part is fine. > > Calls going from the one with dynamic to the static one goes fine. > > Call the other way results now in: > > Failed to authenticate user "1101" <sip:mydk@xxx.xxx.xxx.xxx>At which server?> 1101 is a SIP phone authenticated at the static server. All sip entries > have "canreinvite=no". Two days ago this was working fine. >Yes, there's been quite a lot of changes to SIP registration and authentication. So SIP calls from a user reigstred at the static server to an extension on the dynamic server doesn't work? Is this the setup: <SIP phone 1101> -> SIP CALL -> <Static server> -> SIP CALL -> <dynamic server> ?..and the dynamic server is registred with the static server? Please add a SIP debug of the call so we can see what happens, who refuses what call. /O
Just updated from latest CVS and works like before :-) jo Julian Pawlowski wrote:> The failure has just been fixed as I saw in mantis: > > http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738 > > Thanks a lot! ;D > > > Regards > > Julian Pawlowski > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Well - it turns out it was related to a bug that was introduced on May 24th sometimes but was solved yesterday :) Check out bug 1738.> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Julian > Pawlowski > Sent: 28 May 2004 19:41 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] SIP Changes??? > > > Hi Lars, > > I met the same problems yesterday and even posted it to the list. > Unfortunately nobody answered yet. > > Is it so clear to solve that no one is willing to help us? :-/ > > > Regards, > > Julian Pawlowski > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >