[Asterisk-Users] Can't pickup call when dialing *8 extension

hgaillac-sip at yahoo.fr hgaillac-sip at yahoo.fr
Sat Dec 17 15:48:54 MST 2005


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 at nxs.yi.org:5050 SIP/2.0
Via: SIP/2.0/UDP
80.119.8.167;branch=z9hG4bKe1bb.87855e92.0
From: "alice" <sip:85 at nxs.yi.org>;tag=AF3B88E-55239161
Call-ID: b16b7b62-c85b30e0-5fdbcb3b at 192.168.0.20
To: <sip:*8 at 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 at 192.168.0.20'
    -- Nobody picked up in 10000 ms
Reliably Transmitting (NAT) to 80.119.8.167:5060:
CANCEL sip:86 at 192.168.0.21 SIP/2.0
Via: SIP/2.0/UDP
80.119.8.167:5050;branch=z9hG4bK60e70916;rport
From: "alice"
<sip:84 at 80.119.8.167:5050>;tag=as7cefba23
To: <sip:86 at 192.168.0.21>
Contact: <sip:84 at 80.119.8.167:5050>
Call-ID: 50b2bf516e9f43a5415036b700b0e075 at 80.119.8.167
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0



	

	
		
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