thr3ads.net - asterisk users - [Asterisk-Users] cannot disconnect by callee at first in SIP case [Mar 2003]

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Masakazu Nakano

2003-Mar-01 05:29 UTC

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

sorry, this problem is fixed by myself.

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

---

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)

---
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)

---

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=4dad5671To: "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=4dad5671To: "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=4dad5671To: "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


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

Mark Spencer

2003-Mar-01 10:22 UTC

head link

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

Make sure you're using very latest CVS.  There was a bug that crept in,
where we weren't incrementing the sequence number of our bye.  Does anyone
know what the *correct* rule is for when you do increment on a BYE (or on
a CANCEL) and when you don't?

Mark

On Sat, 1 Mar 2003, Masakazu Nakano wrote:
>
> sorry, this problem is fixed by myself.
>
> we must need set 'canreinvite=no' each user.
>
> ---
>
> 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)
>
> ---
> 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)
>
> ---
>
> 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
>
>
> ---
> Masakazu Nakano as mack at irc
> http://www.dairiten.com:81/
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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