[asterisk-users] Problem When Using Polycom with 2 Lines
Hi, Has anyone had a problem setting up two registrations (on the same Asterisk server) on one Polycom phone? When the user tries to make a call on the 2nd line, it works fine. But when they try the first line, the CLI says:- Using INVITE request as basis request - 9f5fe9a5-215d0f3a-b2fbe...@192.168.1.138 Found peer client _202' <--- Which is incorrect, it should be client_201. And then: [2010-11-15 10:46:29] WARNING[26082]: chan_sip.c:9063 check_auth: username mismatch, have < client _202>, digest has < client _201> [2010-11-15 10:46:29] NOTICE[26082]: chan_sip.c:15079 handle_request_invite: Failed to authenticate user " client line 1" ;tag=E3231B61-69F3BDD6 The IP and port for client_201 and client _202 are the same. Any ideas? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem When Using Polycom with 2 Lines
On Mon, Nov 15, 2010 at 6:05 AM, Dan Journo wrote: > Hi, > > > > Has anyone had a problem setting up two registrations (on the same Asterisk > server) on one Polycom phone? > > When the user tries to make a call on the 2nd line, it works fine. > > > > But when they try the first line, the CLI says:- > > > > Using INVITE request as basis request - > 9f5fe9a5-215d0f3a-b2fbe...@192.168.1.138 > > Found peer client _202' <--- Which is incorrect, it should be client_201. > > And then: > > [2010-11-15 10:46:29] WARNING[26082]: chan_sip.c:9063 check_auth: username > mismatch, have < client _202>, digest has < client _201> > > [2010-11-15 10:46:29] NOTICE[26082]: chan_sip.c:15079 handle_request_invite: > Failed to authenticate user " client line 1" _...@sipserverip>;tag=E3231B61-69F3BDD6 > > > > The IP and port for client_201 and client _202 are the same. > > > > Any ideas? > > Thanks > > Dan > I have a Polycom 550 with 2 lines registered to Asterisk 1.8. I'm using extensions 2000 and 2001. It worked with Asterisk 1.6.2 as well. I do use FreePBX to configure Asterisk. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem When Using Polycom with 2 Lines
Hi! > Using INVITE request as basis request - > 9f5fe9a5-215d0f3a-b2fbe...@192.168.1.138 Found peer client _202' <--- > Which is incorrect, it should be client_201. > > The IP and port for client_201 and client _202 are the same. In short: Asterisk matches by IP address and assigns the INIVTE to the last entry in sip.conf with that IP. In more detail: When Asterisk receives an incoming SIP call, the SIP Channel Module * first tries to find a [user] section matching the caller name (From: username), * then tries to find a [peer] section matching the caller's IP address. * If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf. Source: http://www.voip-info.org/wiki/view/Asterisk+SIP+channels "As of Asterisk 1.2, there is no reason to actually use 'user' entries any more at all; you can use 'type=peer' for everything and the behavior will be much more consistent. All configuration options supported under 'type=user' are also supported under 'type=peer'. The difference between friend and peer is the same as defining _both_ a user and peer, since that is what 'type=friend' does internally. The only benefit of type=user is when you _want_ to match on username regardless of IP the calls originate from. If the peer is registering to you, you don't need it. If they are on a fixed IP, you don't need it. 'type=peer' is _never_ matched on username for incoming calls, only matched on IP address/port number (unless you use insecure=port or higher)." Source: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
