Bonjour a tous.
J'ai un problème pour fermer un channel que je vois quand je fais un "sip show channelstats" :
Code:
#*CLI> sip show channelstats
Peer             Call ID      Duration Recv: Pack  Lost       (     %) Jitter Send: Pack  Lost       (     %) Jitter
192.168.246.100  af05c0da19d           0000000001  0000000000 ( 0.00%) 000000 0000000003  0000000000 ( 0.00%) 000000
1 active SIP channels
Je précise que je ne peux pas le raccrocher en faisant un "channel request hangup".
Le tel est derrière du NAT.
Code:
#*CLI> sip show channel af05c0da19d6b908

  * SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                af05c0da19d6b908
  Owner channel ID:       <none>
  Our Codec Capability:   4
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   4
  Joint Codec Capability:   4
  Format:                 0x0 (nothing)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.246.100:5060
  Received Address:       <public IP of peer>:44228
  SIP Transfer mode:      open
  NAT Support:            Always
  Audio IP:               <server IP> (local)
  Our Tag:                as610a816c
  Their Tag:              93652e8077
  SIP User agent:         Aastra 57i/2.6.0.66
  Username:               <username>
  Peername:               <username>
  Original uri:           sip:<username>@192.168.246.100:5060
  Caller-ID:              100
  Need Destroy:           No
  Last Message:           Tx: INVITE
  Promiscuous Redir:      No
  Route:                  sip:<username>@192.168.246.100:5060;transport=udp
  DTMF Mode:              rfc2833
  SIP Options:            100rel gruu path replaces replace timer 
  Session-Timer:          Inactive

Si quelqu'un sait comment faire sans redémarrer asterisk.....

Merci
Rico