[Asterisk-Users] cannot disconnect by callee at first in SIP case

[Masakazu Nakano]

sorry, this problem is fixed by myself.

we must need set 'canreinvite=no' each user.

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I'm try to discconect a call with SIP.

when caller make a call, 'show channels' result is following.
mack*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
  SIP/mack-1bfc  (default                 1   ) Ringing AppDial       (Outgoing Line)
 SIP/mack2-8c2f  (default    110          1   )    Ring Dial          SIP/mack
2 active channel(s)

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and caller maked a call, 'show channels' result is following.
mack*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
  SIP/mack-1bfc  (default                 1   )      Up Bridged Call  SIP/mack2-8c2f
 SIP/mack2-8c2f  (default    110          1   )      Up Dial          SIP/mack
2 active channel(s)

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and callee disconnect this call,  'show channels' result is following.
mack*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         Data
0 active channel(s)

but callee still displayed 'Connected with' ( in snom100 case )
and transmit BYE to caller in 'sip debug' result.
and next send INVITE by asterisk again in following under.

why???

  == Spawn extension (default, 110, 1) exited non-zero on 'SIP/mack2-eba6'
XXX Need to handle Retransmitting XXX:
BYE sip:mack2 at 192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=76bf5f20
>From: <sip:110 at 192.168.0.1;user=phone>;tag=4dad5671
To: "mack2" <sip:mack2 at 192.168.0.1>;tag=4wvwq4r7lt
Call-ID: 3e6096da795e-fc52831snxub at 210.194.204.16
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

 to 192.168.0.14:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=76bf5f20;rport=5060
>From: <sip:110 at 192.168.0.1;user=phone>;tag=4dad5671
To: "mack2" <sip:mack2 at 192.168.0.1>;tag=4wvwq4r7lt
Call-ID: 3e6096da795e-fc52831snxub at 210.194.204.16
CSeq: 103 INVITE
Session-Expires: 3600
User-Agent: snom100-1.15e
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE
Supported: timer, 100rel, replaces
Contact: <sip:mack2 at 192.168.0.14:5060;transport=udp;line=1>
Content-Length: 242

v=0
o=root 30701 30701 IN IP4 192.168.0.14
s=SIP Call
c=IN IP4 192.168.0.14
t=0 0
m=audio 10000 RTP/AVP 3 101
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=x-private:192.168.0.14:10000 210.194.204.16:46930

13 headers, 10 lines
Message is INVITE
XXX Need to handle Retransmitting XXX:
ACK sip:mack2 at 192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=76bf5f20
>From: <sip:110 at 192.168.0.1;user=phone>;tag=4dad5671
To: "mack2" <sip:mack2 at 192.168.0.1>;tag=4wvwq4r7lt
Call-ID: 3e6096da795e-fc52831snxub at 210.194.204.16
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 to 192.168.0.14:5060


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Masakazu Nakano as mack at irc
http://www.dairiten.com:81/