Got SIP response 486 'Busy Here' and Aastra 620d phone

[marvinhorst]

When dialing from fop2 the sip stack on the Astra 620d sip dect phones rejects calls if the caller id number matches the sip channel. So when asterisk originates the initial leg of the call I get the 486 response.

From sip debug:
This doesn't work....
From: "Brian Witmer" sip:135[/color:1yy6ve7s]@192.168.16.207;tag=as7e879e22

This does....
From: "Brian Witmer" sip:web[/color:1yy6ve7s]@192.168.16.207;tag=as60489c59

My sip.conf has this:

[135](aastra,paulb)
username=135
mailbox=135@paulb
callerid="Brian Witmer" <135>

I know that it isn't recommended procedure to name the sip channel with the extension number, but these phones display the sip account number on screen and it's a handy way to display the extension number on the limited screen space.

Is there any way to get fop2 to not use the extension number as the caller id number when generating the originate command??

my buttons.cfg

[SIP/135]
group=PaulB
label=Brian Witmer
extension=135
context=internal
mailbox=135@paulb
type=extension

[nicolas]

Currently, there is no way to change the callerid.. you may want to try some "hacks" that might work: for example, you can create a special context for fop2 extensions like this very crude example:

[fop-extensions]
exten => 99135,1,Dial(SIP/135)
exten => 9913561,Dial(SIP/136)

So, you have extension 99135 dialing sip/135

Then, in the fop2 button configuration set

[SIP/135]
label=something
type=extension
extension=99135
context=fop-extensions
...

I think that the callerid when originating calls will be 99135 and perhaps your Aastra phone will be happy. I really do not know if it will actually work, it is just a wild idea.

Best regards,

[Cyril]

Hi Nicolas,

Do you have any idea for this pb ?

My configuration :

AASTRA RFP L32 IP OpenMobility Manager v1.8.9 with last update
15 DECT 142D AASTRA with last update
10 TB30 THOMSON with last update
ELASTIX 2.0.0-57
FOP2 registred fop2-2.22-centos5

With 10 Thomson works fine but with AASTRA DECT i have the same problem :
== Using SIP RTP CoS mark 5
-- Got SIP response 486 "Busy Here" back from X.X.X.X
-- Executing [failed@from-internal:1] Macro("OutgoingSpoolFailed", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("OutgoingSpoolFailed", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("OutgoingSpoolFailed", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("OutgoingSpoolFailed", "1?Set(REALCALLERIDNUM=)") in new stack

Do you have any idea ?

Best regards,

Cyril

[nicolas]

Hi,

I have already proposed a solution. Do not use the same extension in your fop2 button configuration than your phone's.. add a prefix to it, and use a custom dialplan to still call your device removing that prefix. Perhaps it will work that way. A phone should not reject a call because it comes from the same callerid as itself.