[asterisk-dev] Many sip dialog/ opened channels.

[Catalin S.]

Hello,

I'm using asterisk with 84 extensions (aprox 45 always connected). When i
look to the opened channels i sow many channels opened without reason even i
don't have any active calls.
Is there someone else that en-counted the same problem? Is there any fix to
this bug? I have the following settings:


Global Settings:
----------------
  UDP Bindaddress:        [::]:5060
  ** Additional Info:
     [::] may include IPv4 in addition to IPv6, if such a feature is enabled
in the OS.
  TCP SIP Bindaddress:    [::]:5060
  TLS SIP Bindaddress:    Disabled
  Videosupport:           Yes
  Textsupport:            Yes
  Ignore SDP sess. ver.:  Yes
  AutoCreate Peer:        No
  Match Auth Username:    No
  Allow unknown access:   No
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   Yes
  SIP domain support:     Yes
  Realm. auth:            No
  Our auth realm          sip.someprovider.info
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   Yes
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             asterisk
  SDP Session Name:       Asterisk PBX 1.8.7.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Legacy userfield parse: No
  Caller ID:              asterisk
  From: Domain:           sip.someprovider.info
  Record SIP history:     On
  Call Events:            On
  Auth. Failure Events:   Off
  T.38 support:           Yes
  T.38 EC mode:           FEC
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          5000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  IP ToS RTP text:        AF41
  802.1p CoS SIP:         3
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   4
  802.1p CoS RTP text:    3
  Jitterbuffer enabled:   Yes
  Jitterbuffer forced:    No
  Jitterbuffer max size:  300
  Jitterbuffer resync:    1000
  Jitterbuffer impl:      fixed
  Jitterbuffer log:       No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          5

Global Signalling Settings:
---------------------------
  Codecs:                 0xe (gsm|ulaw|alaw)
  Codec Order:            ulaw:20,alaw:20,gsm:20
  Relax DTMF:             No
  RFC2833 Compensation:   Yes
  Symmetric RTP:          No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            120
  RTP Hold Timeout:       600
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   No
  Reg. min duration       30 secs
  Reg. max duration:      80 secs
  Reg. default duration:  1800 secs
  Outbound reg. timeout:  30 secs
  Outbound reg. attempts: 5
  Notify ringing state:   Yes
    Include CID:          Yes
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           Yes
  Outb. proxy:            <not set>
  Session Timers:         Refuse
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                default
  Force rport:            No
  DTMF:                   rfc2833
  Qualify:                500
  Use ClientCode:         No
  Progress inband:        Yes
  Language:               en
  MOH Interpret:          default
  MOH Suggest:            default
  Voice Mail Extension:   voicemail

and the opened channels:

rr-de*CLI> sip show channels
Peer             User/ANR         Call ID          Format           Hold
Last Message    Expiry     Peer
6.6.13.17   (None)           000750d5-411d00  0x0 (nothing)    No       Rx:
REGISTER               <guest>
6.1.13.17   (None)           7de7064b-6f9f69  0x0 (nothing)      No
Rx: REGISTER               <guest>
6.1.18.13   (None)           08a2e79c7f13b73  0x0 (nothing)    No       Rx:
REGISTER               <guest>
1.2.12.23   (None)           000dbcd9-39db00  0x0 (nothing)    No       Rx:
REGISTER               <guest>
8.6.13.17   (None)           000750d5-411d00  0x0 (nothing)    No       Rx:
REGISTER               <guest>
8.1.13.17   (None)           ca30cc15-d93e4d  0x0 (nothing)    No       Rx:
REGISTER               <guest>
6.1.12.17   (None)           226b901d-4bff19  0x0 (nothing)      No
Rx: REGISTER               <guest>
9.1.12.20   (None)           2474013819 at 192_  0x0 (nothing)  No       Rx:
REGISTER               <guest>
2.1.14.10   (None)           d1bb5072-b6ebcd  0x0 (nothing)    No       Rx:
REGISTER               <guest>
xxxxx
xxx
xxxx
8.1.13.17   (None)           ca30cc15-d93e4d  0x0 (nothing)    No       Rx:
REGISTER               <guest>
6.1.12.17   (None)           226b901d-4bff19  0x0 (nothing)      No
Rx: REGISTER               <guest>
9.1.12.20   (None)           2474013819 at 192_  0x0 (nothing)  No       Rx:
REGISTER               <guest>
2.1.14.10   (None)           d1bb5072-b6ebcd  0x0 (nothing)    No       Rx:
REGISTER               <guest>

*4423 active SIP dialogs*
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