On 04/18/2011 05:33 PM, Warren Selby wrote:
On Mon, Apr 18, 2011 at 4:54 AM, Jonas Kellens <[email protected] <mailto:[email protected]>> wrote:

    Hello list,

    I have in sip.conf :


<snip>

    So are my settings wrong ?


What does sip show settings look like from the CLI?

vps*CLI> sip show settings


Global Settings:
----------------
  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           Disabled
  TLS SIP Port:           Disabled
  Videosupport:           Yes
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        No
  Match Auth Username:    No
  Allow unknown access:   No
  Allow subscriptions:    Yes
  Allow overlap dialing:  No
  Allow promsic. redir:   No
  Enable call counters:   Yes
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          domain.be
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 1.6.2.16.1
  SDP Session Name:       Asterisk PBX 1.6.2.16.1
  SDP Owner Name:         owner
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              0
  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:           Enabled
  Qualify Freq :          120000 ms

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         3
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   4
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No
  Jitterbuffer forced:    No
  Jitterbuffer max size:  -1
  Jitterbuffer resync:    -1
  Jitterbuffer impl:
  Jitterbuffer log:       No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost: <none>
  Externip:               0.0.0.0:0
  Externrefresh:          10
  STUN server:            0.0.0.0:0

Global Signalling Settings:
---------------------------
  Codecs:                 0x28090a (gsm|alaw|g726|g729|h263|h264)
  Codec Order:            alaw:20,g726:20,g729:20,gsm:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            60
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      60 secs
  Reg. default duration:  300 secs
  Outbound reg. timeout:  240 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      Yes
  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

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:      UDP
  Context:                default
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               nl
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk
  Forward Detected Loops: Yes

Realtime SIP Settings:
----------------------
  Realtime Peers:         Yes
  Realtime Regs:          No
  Cache Friends:          Yes
  Update:                 Yes
  Ignore Reg. Expire:     No
  Save sys. name:         No
  Auto Clear:             120 (Disabled)

--
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