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

[Drew Gibson]

Thanks Andrew, I understand the issue now.

Removing "insecure=very" allows the Grandstream phones to work, they 
register separate lines on separate ports (eg Line 1=5060, Line 2=5062, 
etc).

Unfortunately I cannot find a port setting for the Aastra 480i, I shall 
get on their case.

regards,

Drew


Andrew Joakimsen wrote:
> ;------------------------------------------------------------------------------ 
>
> ; 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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-- 
Drew Gibson

Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com