Reply To: Trunk Dial Out

VitalPBX Community Support General Discussion Trunk Dial Out Reply To: Trunk Dial Out

    Here are the show sip settings:

    Global Settings:
    UDP Bindaddress:
    TCP SIP Bindaddress:
    TLS SIP Bindaddress:
    RTP Bindaddress: Disabled
    Videosupport: Yes
    Textsupport: No
    Ignore SDP sess. ver.: No
    AutoCreate Peer: Off
    Match Auth Username: No
    Allow unknown access: Yes
    Allow subscriptions: Yes
    Allow overlap dialing: Yes
    Allow promisc. redir: No
    Enable call counters: Yes
    SIP domain support: No
    Path 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: VitalPBX
    SDP Session Name: Asterisk PBX 17.7.0
    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
    Auth. Failure Events: Off
    T.38 support: No
    T.38 EC mode: Unknown
    T.38 MaxDtgrm: 4294967295
    SIP realtime: Disabled
    Qualify Freq : 60000 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: No

    Network Settings:
    SIP address remapping: Enabled using externhost
    Externrefresh: 10

    Global Signalling Settings:
    Codecs: (ulaw|alaw|g729)
    Relax DTMF: No
    RFC2833 Compensation: No
    Symmetric RTP: Yes
    Compact SIP headers: No
    RTP Keepalive: 0 (Disabled)
    RTP Timeout: 30
    RTP Hold Timeout: 300
    MWI NOTIFY mime type: application/simple-message-summary
    DNS SRV lookup: No
    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:No
    Notify ringing state: Yes
    Include CID: Yes
    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: sip-default
    Record on feature: one_touch_
    Record off feature: one_touch_
    Force rport: Yes
    DTMF: rfc2833
    Qualify: 0
    Keepalive: 0
    Use ClientCode: No
    Progress inband: No
    Language: en
    Tone zone: us
    MOH Interpret: default
    MOH Suggest:
    Voice Mail Extension: *97
    RTCP Multiplexing: No


    <— SIP read from UDP: —>