[asterisk-users] Multi-line phones - Asterisk uses wrong callerid

[Andrew Joakimsen]

;------------------------------------------------------------------------------
; 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 <drew at oanda.com> 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
>
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