Summary:	ASTERISK-04924: even though type=user is set in sip.conf incoming connection context finding fails
Reporter:	barthek (barthek)	Labels:
Date Opened:	2005-08-29 09:57:57	Date Closed:	2011-06-07 14:09:59
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	i have registered with upstream SIP provider and only want to accept calls from it. when i have type=user set in sip.conf asterisk wont accept incoming call and spits: Using INVITE request as basis request - A6E012BB-17D211DA-B072801A-D5BC1E91@83.238.255.146 Sending to 83.238.255.132 : 5060 (non-NAT) Found no matching peer or user for '83.238.255.132:5060' when i change type to friend or peer everything works fine.
Comments:	By: Olle Johansson (oej) 2005-08-29 10:24:10 This is not a bug, it is a configuration issue. Asterisk identifies callers and matches with users based on the From: user header, which changes for every call from your service provider. You have to match on IP to a peer. There are several configuration examples out there for you to read and copy from, search on "Asterisk Broadvoice" as an example. Closing this issue report, since it is not a bug. By: barthek (barthek) 2005-08-29 10:34:04 just a note: i think documentation and provided examples are misleading. in a few places i have found information that if you want outgoing calls placed then you should set type=peer and if incoming, type=user. Set it to type=friend if you need both-way calls. as a novice user i thought type directive applies to way in which you want the calls to be matched. take a look at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20type and correct me if i got it wrong. i kindly ask for documentation clarification :) By: Olle Johansson (oej) 2005-08-29 10:41:19 There are a lot of docs out there that says that we match incoming calls on peers in some cases. Please read available docs... By: Olle Johansson (oej) 2005-08-29 10:41:53 --- This is not a support channel. Find us in #asterisk-dev or #asterisk-bugs on IRC.