Hello,<div><br></div><div>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.</div><div>
Is there someone else that en-counted the same problem? Is there any fix to this bug? I have the following settings:</div><div><br></div><div><div><br></div><div>Global Settings:</div><div>----------------</div><div> UDP Bindaddress: [::]:5060</div>
<div> ** Additional Info:</div><div> [::] may include IPv4 in addition to IPv6, if such a feature is enabled in the OS.</div><div> TCP SIP Bindaddress: [::]:5060</div><div> TLS SIP Bindaddress: Disabled</div>
<div> Videosupport: Yes</div><div> Textsupport: Yes</div><div> Ignore SDP sess. ver.: Yes</div><div> AutoCreate Peer: No</div><div> Match Auth Username: No</div><div> Allow unknown access: No</div>
<div> Allow subscriptions: Yes</div><div> Allow overlap dialing: Yes</div><div> Allow promisc. redir: No</div><div> Enable call counters: Yes</div><div> SIP domain support: Yes</div><div> Realm. auth: No</div>
<div> Our auth realm <a href="http://sip.someprovider.info">sip.someprovider.info</a></div><div> Use domains as realms: No</div><div> Call to non-local dom.: Yes</div><div> URI user is phone no: Yes</div><div>
Always auth rejects: Yes</div><div> Direct RTP setup: No</div><div> User Agent: asterisk</div><div> SDP Session Name: Asterisk PBX 1.8.7.0</div><div> SDP Owner Name: root</div><div>
Reg. context: (not set)</div><div> Regexten on Qualify: No</div><div> Legacy userfield parse: No</div><div> Caller ID: asterisk</div><div> From: Domain: <a href="http://sip.someprovider.info">sip.someprovider.info</a></div>
<div> Record SIP history: On</div><div> Call Events: On</div><div> Auth. Failure Events: Off</div><div> T.38 support: Yes</div><div> T.38 EC mode: FEC</div><div> T.38 MaxDtgrm: -1</div>
<div> SIP realtime: Disabled</div><div> Qualify Freq : 5000 ms</div><div> Q.850 Reason header: No</div><div> Store SIP_CAUSE: No</div><div><br></div><div>Network QoS Settings:</div><div>---------------------------</div>
<div> IP ToS SIP: CS3</div><div> IP ToS RTP audio: EF</div><div> IP ToS RTP video: AF41</div><div> IP ToS RTP text: AF41</div><div> 802.1p CoS SIP: 3</div><div> 802.1p CoS RTP audio: 5</div>
<div> 802.1p CoS RTP video: 4</div><div> 802.1p CoS RTP text: 3</div><div> Jitterbuffer enabled: Yes</div><div> Jitterbuffer forced: No</div><div> Jitterbuffer max size: 300</div><div> Jitterbuffer resync: 1000</div>
<div> Jitterbuffer impl: fixed</div><div> Jitterbuffer log: No</div><div><br></div><div>Network Settings:</div><div>---------------------------</div><div> SIP address remapping: Disabled, no localnet list</div>
<div> Externhost: <none></div><div> Externaddr: (null)</div><div> Externrefresh: 5</div><div><br></div><div>Global Signalling Settings:</div><div>---------------------------</div>
<div> Codecs: 0xe (gsm|ulaw|alaw)</div><div> Codec Order: ulaw:20,alaw:20,gsm:20</div><div> Relax DTMF: No</div><div> RFC2833 Compensation: Yes</div><div> Symmetric RTP: No</div>
<div> Compact SIP headers: No</div><div> RTP Keepalive: 0 (Disabled)</div><div> RTP Timeout: 120 </div><div> RTP Hold Timeout: 600 </div><div> MWI NOTIFY mime type: application/simple-message-summary</div>
<div> DNS SRV lookup: No</div><div> Pedantic SIP support: No</div><div> Reg. min duration 30 secs</div><div> Reg. max duration: 80 secs</div><div> Reg. default duration: 1800 secs</div><div> Outbound reg. timeout: 30 secs</div>
<div> Outbound reg. attempts: 5</div><div> Notify ringing state: Yes</div><div> Include CID: Yes</div><div> Notify hold state: No</div><div> SIP Transfer mode: open</div><div> Max Call Bitrate: 384 kbps</div>
<div> Auto-Framing: Yes</div><div> Outb. proxy: <not set> </div><div> Session Timers: Refuse</div><div> Session Refresher: uas</div><div> Session Expires: 1800 secs</div>
<div> Session Min-SE: 90 secs</div><div> Timer T1: 500</div><div> Timer T1 minimum: 100</div><div> Timer B: 32000</div><div> No premature media: Yes</div><div> Max forwards: 70</div>
<div><br></div><div>Default Settings:</div><div>-----------------</div><div> Allowed transports: UDP</div><div> Outbound transport: UDP</div><div> Context: default</div><div> Force rport: No</div>
<div> DTMF: rfc2833</div><div> Qualify: 500</div><div> Use ClientCode: No</div><div> Progress inband: Yes</div><div> Language: en</div><div> MOH Interpret: default</div>
<div> MOH Suggest: default</div><div> Voice Mail Extension: voicemail</div></div><div><br></div><div>and the opened channels:</div><div><br></div><div><div>rr-de*CLI> sip show channels</div><div>Peer User/ANR Call ID Format Hold Last Message Expiry Peer </div>
</div><div><div>6.6.13.17 (None) 000750d5-411d00 0x0 (nothing) No Rx: REGISTER <guest> </div><div>6.1.13.17 (None) 7de7064b-6f9f69 0x0 (nothing) No Rx: REGISTER <guest> </div>
<div>6.1.18.13 (None) 08a2e79c7f13b73 0x0 (nothing) No Rx: REGISTER <guest> </div><div>1.2.12.23 (None) 000dbcd9-39db00 0x0 (nothing) No Rx: REGISTER <guest> </div>
<div>8.6.13.17 (None) 000750d5-411d00 0x0 (nothing) No Rx: REGISTER <guest> </div><div>8.1.13.17 (None) ca30cc15-d93e4d 0x0 (nothing) No Rx: REGISTER <guest> </div>
<div>6.1.12.17 (None) 226b901d-4bff19 0x0 (nothing) No Rx: REGISTER <guest> </div><div>9.1.12.20 (None) 2474013819@192_ 0x0 (nothing) No Rx: REGISTER <guest> </div>
<div>2.1.14.10 (None) d1bb5072-b6ebcd 0x0 (nothing) No Rx: REGISTER <guest></div><div>xxxxx</div><div>xxx</div><div>xxxx</div><div>8.1.13.17 (None) ca30cc15-d93e4d 0x0 (nothing) No Rx: REGISTER <guest> </div>
<div>6.1.12.17 (None) 226b901d-4bff19 0x0 (nothing) No Rx: REGISTER <guest> </div><div>9.1.12.20 (None) 2474013819@192_ 0x0 (nothing) No Rx: REGISTER <guest> </div>
<div>2.1.14.10 (None) d1bb5072-b6ebcd 0x0 (nothing) No Rx: REGISTER <guest></div><div><br></div><div><b><font class="Apple-style-span" color="#ff0000">4423 active SIP dialogs</font></b></div>
</div><div><br></div>