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* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111013/259fdab7/attachment.htm>