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please help me about my sip.conf http://forum.asterisk2billing.org/viewtopic.php?f=33&t=11924 |
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Author: | ziaabass [ Mon Nov 30, 2015 7:51 pm ] |
Post subject: | please help me about my sip.conf |
Code: *CLI> sip show settings Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: Disabled TLS SIP Bindaddress: Disabled Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: Off Match Auth Username: Yes Allow unknown access: No Allow subscriptions: Yes Allow overlap dialing: Yes Allow promisc. redir: No Enable call counters: No SIP domain support: No Realm. auth: No Our auth realm asterisk Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk PBX 11.7.0~dfsg-1ubuntu1 SDP Session Name: Asterisk PBX 11.7.0~dfsg-1ubuntu1 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Trust RPID: No Send RPID: No Legacy userfield parse: No Send Diversion: Yes Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 60000 ms Q.850 Reason header: No Store SIP_CAUSE: No Network QoS Settings: --------------------------- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: No Network Settings: --------------------------- SIP address remapping: Disabled, no localnet list Externhost: <none> Externaddr: (null) Externrefresh: 10 Global Signalling Settings: --------------------------- Codecs: (gsm|g729) Codec Order: g729:20,gsm:20 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Sub. min duration 60 secs Sub. max duration: 3600 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Outbound reg. retry 403:0 Notify ringing state: Yes Include CID: No Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: <not set> Session Timers: Accept 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: a2billing Record on feature: automon Record off feature: automon Force rport: Auto (No) DTMF: rfc2833 Qualify: 0 Keepalive: 0 Use ClientCode: No Progress inband: Never Language: Tone zone: <Not set> MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk i have a only sip clients so witch option is must i configure |
Author: | ziaabass [ Sat Dec 05, 2015 5:10 pm ] |
Post subject: | Re: please help me about my sip.conf |
no one have a discus on this topic please share your good config hare |
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