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 <[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
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--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
www.oanda.com
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