;------------------------------------------------------------------------------ ; Definitions of locally connected SIP phones ; ; type = user a device that authenticates to us by "from" field to place calls ; type = peer a device we place calls to or that calls us and we match by host ; type = friend two configurations (peer+user) in one ;
Thus if you have two "peers" using the same IP address AND port it will probably match. First try to remove insecure=very from your configuration file, that alone might resolve it. If not you need to insure that each line gets its own port. On 3/28/07, Drew Gibson <[EMAIL PROTECTED]> wrote:
I have some phones (and an ATA) that are shared between two users who each have separate voicemail but they are not behaving as desired nor expected. Incoming calls show up on the correct lines. Calls originating from the device are seen, at the terminating device, as coming from the account listed last in sip.conf, regardless of the line selected. This creates three main issues I would like to resolve:- 1. The person called sees the wrong callerid 2. The CDR records the call against the wrong account 3. Picking up voicemail requires multiple extra steps Is there a way around this?? Scenario:- Phone 1 has three lines 101, 102, 103 Phone 2 has 1 line 202 User 1 selects line 101 at Phone 1 and dials 202 (to Phone 2) User 2 at Phone 2 sees call coming from extension 103 instead of 101 With 'sip debug' enabled at the console, I see an INVITE issued (on the Phone 1 to Asterisk leg) from the correct extension, 101, to 202 but the call leg from Asterisk to Phone 202 shows an INVITE from 103 to 202. 103 happens to be the last listed in sip.conf and the first listed in 'sip show peers' (I have confirmed that this is dependent on the order in the conf file, not numeric order) sip.conf :- [general] port = 5060 bindaddr = 0.0.0.0 pedantic = no autocreatepeer = no context = sip registertimeout=20 localnet = 10.10.10.0/255.255.255.0 srvlookup = yes tos=0xb8 rtptimeout=300 rtpholdtimeout=1800 maxexpirey=3600 defaultexpirey=1200 [sip-101] ; Aastra 480i phones for general office type=peer insecure=very disallow=all allow=ulaw allow=alaw host=dynamic dtmfmode=auto canreinvite=no context=office-dial qualify=yes username=101 secret=xxxxxx mailbox=101 callerid="User 1" <101> sip show peers :- 103/103 10.10.10.181 D 5060 OK (157 ms) 102/102 10.10.10.181 D 5060 OK (159 ms) 202/202 10.10.10.184 D 5060 OK (4 ms) 101/101 10.10.10.181 D 5060 OK (160 ms) Asterisk 1.2.15 Phones tested:- Aastra 480i, Grandstream GXP2000, Grandstream HT-386 ATA -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com _______________________________________________ --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
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