asterisk users - Dec 2005 - Can't pickup call when dialing *8 extension

Hello,

I added callgroup=1 and pickupgroup=1 for sip channels
however I can't pickup a call (see below ) between sip
phones when i dial *8 .

May I have to add app_pickup to solve this problem.
I use asterisk-1.2

Regards
Harry


serveur1*CLI>
<-- SIP read from 80.119.8.167:5060:
ACK sip:*8@nxs.yi.org:5050 SIP/2.0
Via: SIP/2.0/UDP
80.119.8.167;branch=z9hG4bKe1bb.87855e92.0
From: "alice" <sip:85@nxs.yi.org>;tag=AF3B88E-55239161
Call-ID: b16b7b62-c85b30e0-5fdbcb3b@192.168.0.20
To: <sip:*8@nxs.yi.org>;tag=as543ba455
CSeq: 2 ACK
User-Agent: Sip EXpress router(0.9.4 (i386/linux))
Content-Length: 0

--- (8 headers 0 lines)---
Destroying call
'b16b7b62-c85b30e0-5fdbcb3b@192.168.0.20'
    -- Nobody picked up in 10000 ms
Reliably Transmitting (NAT) to 80.119.8.167:5060:
CANCEL sip:86@192.168.0.21 SIP/2.0
Via: SIP/2.0/UDP
80.119.8.167:5050;branch=z9hG4bK60e70916;rport
From: "alice"
<sip:84@80.119.8.167:5050>;tag=as7cefba23
To: <sip:86@192.168.0.21>
Contact: <sip:84@80.119.8.167:5050>
Call-ID: 50b2bf516e9f43a5415036b700b0e075@80.119.8.167
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



	

	
		
___________________________________________________________________________ 
Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs
exceptionnels pour appeler la France et l'international.
T?l?chargez sur http://fr.messenger.yahoo.com

You might have to use *8#. At least I do with my 7960.

------------------------
> I added callgroup=1 and pickupgroup=1 for sip channels
> however I can't pickup a call (see below ) between sip
> phones when i dial *8 .
> 
> May I have to add app_pickup to solve this problem.
> I use asterisk-1.2
> 
> Regards
> Harry
> 
> 
> serveur1*CLI>
> <-- SIP read from 80.119.8.167:5060:
> ACK sip:*8@nxs.yi.org:5050 SIP/2.0
> Via: SIP/2.0/UDP
> 80.119.8.167;branch=z9hG4bKe1bb.87855e92.0
> From: "alice" <sip:85@nxs.yi.org>;tag=AF3B88E-55239161
> Call-ID: b16b7b62-c85b30e0-5fdbcb3b@192.168.0.20
> To: <sip:*8@nxs.yi.org>;tag=as543ba455
> CSeq: 2 ACK
> User-Agent: Sip EXpress router(0.9.4 (i386/linux))
> Content-Length: 0
> 
> --- (8 headers 0 lines)---
> Destroying call
> 'b16b7b62-c85b30e0-5fdbcb3b@192.168.0.20'
>     -- Nobody picked up in 10000 ms
> Reliably Transmitting (NAT) to 80.119.8.167:5060:
> CANCEL sip:86@192.168.0.21 SIP/2.0
> Via: SIP/2.0/UDP
> 80.119.8.167:5050;branch=z9hG4bK60e70916;rport
> From: "alice"
> <sip:84@80.119.8.167:5050>;tag=as7cefba23
> To: <sip:86@192.168.0.21>
> Contact: <sip:84@80.119.8.167:5050>
> Call-ID: 50b2bf516e9f43a5415036b700b0e075@80.119.8.167
> CSeq: 102 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> 
> 
> 
> 	
> 
> 	
> 		
> ___________________________________________________________________________
> Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs
exceptionnels pour appeler la France et l'international.
> T?l?chargez sur http://fr.messenger.yahoo.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
> 
---------------End of Original Message-----------------