Summary:	ASTERISK-00761: omitting username in sip.conf don't let to call a cisco phone until it registers again
Reporter:	Matteo Brancaleoni (mbrancaleoni)	Labels:
Date Opened:	2004-01-08 13:31:56.000-0600	Date Closed:	2008-01-15 14:40:34.000-0600
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) bug_767.diff.txt
Description:	the username part in sip.conf seems pretty not useful, cause the auth name is taken from the [blah] part. So omitting username=blah is obvious sometime. btw, if omitting it, I noticed that you can't call a cisco phone until it registers again, since asterisk sends out invites without the blah@ part in the message, like that: INVITE sip:192.168.x.x SIP/2.0 only when the phone registers again, we have a correct INVITE sip:blah@192.168.x.x SIP/2.0 If username=blah is added to sip.conf, then even if the phone isn't registered (but * knows what IP to call, since is stored into the astdb) we have a correct invite message. so I think we must obsolete the username= part (because auth data is taken from []) and fix chan_sip to use the [blah] for seding out invites & perhaps other messages.
Comments:	By: Brian West (bkw918) 2004-01-09 00:06:42.000-0600 Good idea.. I also see this behavior. By: Matteo Brancaleoni (mbrancaleoni) 2004-01-09 09:54:40.000-0600 now we must decide what do do... A) obsolete username= option and go only with [<name>] part? B) don't take [<name>] part for auth and really use username ? By: Brian West (bkw918) 2004-01-09 09:58:35.000-0600 How about leave it like it is.. and copy[<name>] to username if username isn't defined? bkw By: pliew (pliew) 2004-01-09 23:06:40.000-0600 I agree, copy name to username if it doesn't exist, this way it won't break anything, especially if there are existing dependencies on username like "incominglimit". Paul Ok, I've added a little patch to do just that. Hopefully that helps. edited on: 01-09-04 23:29 By: Olle Johansson (oej) 2004-01-10 02:30:15.000-0600 Please test the new sip channel (bug 0000759) where I changed the behaviour of registration/invites. in this version (yet to be tested by others than me), I save the full contact: header from registrations and use that in later INVITES to those peers. In some cases, like after a restart, Asterisk reverts to the old behaviour until a new registration occurs. This is related to the SNOM problem (bug 0000732) and the reported problem with some middleman NAT boxes. By: Brian West (bkw918) 2004-01-11 15:38:30.000-0600 Fixed in CVS By: Digium Subversion (svnbot) 2008-01-15 14:40:34.000-0600 Repository: asterisk Revision: 1956 U   trunk/channels/chan_sip.c ------------------------------------------------------------------------ r1956 | jeremy | 2008-01-15 14:40:34 -0600 (Tue, 15 Jan 2008) | 2 lines Fix ast-db seeding. Bug ASTERISK-761 ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=1956