[Asterisk-Dev] locked sip channels

[Chee Foong]

Hello,

Please enable sip debug and capture the sip messages of a locked SIP calls
then we can examine further. I had experience this few weeks back and that
happends to me incompatibility of SIP implementation between * and the other
end. I hear a timeout or sort of solution is being planned but I am not
sure.


CCF
  -----Original Message-----
  From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com]On Behalf Of Dov Bigio
  Sent: Thursday, September 01, 2005 02:56
  To: asterisk-dev at lists.digium.com
  Subject: [Asterisk-Dev] locked sip channels



  Hello,

  I am using Asterisk behind SER and connecting to AudioCodes gateway to
link to my legacy PBX.

  When I run "show channels" I get the following active channels

  LV07*CLI> show channels
          Channel  (Context    Extension    Pri )   State Appl.         Data
  SIP/ana.paula.furuya-683a  (01.cobranca              1   ) Ringing AppDial
(Outgoing Line)
  SIP/GW215-b6ee6da8  (macro-ramaisSemVM s            5   )      Up Dial
SIP/ana.paula.furuya|15|tr
          Zap/1-1  (entradazaptel s            1   )      Up Bridged Call
SIP/eliza.silva-99ec
  SIP/eliza.silva-99ec  (01.cobranca 003138914480 5   )      Up Dial
Zap/g1/0213138914480|60
       Agent/5150  (default    s            1   )      Up Bridged Call
SIP/GW212-b7011350
  SIP/diogo.vomero-cbc0  (default    s            1   )      Up (None)
(None)
  SIP/GW212-b7011350  (01.filas.cobranca cobranca     6   )      Up Queue
cobranca|tT|||500000
  SIP/andreagora-43ba  (01.diretoria              1   )      Up Bridged Call
SIP/francisco.zapata-9eae
  SIP/francisco.zapata-9eae  (macro-ramais s            6   )      Up Dial
SIP/andreagora|15|tr
  SIP/marcus.rocha-c07a  (01.administrativo              1   )      Up
Bridged Call  SIP/GW211-b6e43478
  SIP/GW211-b6e43478  (macro-ramais s            6   )      Up Dial
SIP/marcus.rocha|15|tr
  11 active channel(s)

  But when I run "sip show channels" I have a much bigger list.

  LV07*CLI> sip show channels
  Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format
  200.234.206.113  (None)      330b200da51  00101/00012   unknow
  200.234.206.113  (None)      6d264f57c41  00101/00005   unknow
  10.2.0.2         ana.paula.  157f8cb70c8  00102/00000   ulaw
  10.5.0.11        celso.paul  60caaa0d402  00102/00000   unknow
  200.234.206.215  "307" <pab  91072140321  00101/26900680   ulaw
  200.234.206.113  eliza.silv  db087234996  00101/00001   ulaw
  10.2.0.4         diogo.vome  560558f1241  00102/00000   ulaw
  200.234.206.212  "292" <292  17558894989  00101/32199696   ulaw
  10.0.0.7         andreagora  4b2488c71e9  00102/00000   ulaw
  200.234.206.113  francisco.  02541676a54  00101/00001   ulaw
  10.0.0.5         marcus.roc  18043d091bf  00102/00000   ulaw
  200.234.206.211  "498" <498  27022300603  00101/17831348   ulaw
  200.234.206.113  diogo.vome  09daea2a7c4  00101/00003   ulaw
  200.234.206.215  350         5228b91e680  00103/26486668   ulaw
  200.234.206.212  "340" <340  38981259125  00102/31783441   ulaw
  200.234.206.215  350         62a3c2537fd  00103/26365001   ulaw
  200.234.206.212  "343" <343  10871178911  00102/31664598   ulaw
  200.234.206.215  350         589da13441d  00103/26345971   ulaw
  200.234.206.212  "342" <342  31456118051  00102/31644425   ulaw

  By examining this channels through Flash Operator Panel, it says that I
have calls that are lasting for hours (those extra channels listed in sip
show channels), which is not true.

  Is there a way to hang up those invalid channels?

  My Asterisk (1.0.9) locks from time to time with, being overloaded with
never-closing channels...

  Thank you!
  Dov
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