[asterisk-bugs] [JIRA] (ASTERISK-24762) not able to make outbound call from Asterisk/1.8.13.1

[Ashish (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24762?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224782#comment-224782 ] 

Ashish commented on ASTERISK-24762:
-----------------------------------

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:        No
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     Yes
  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 1.8.13.1~dfsg1-3+deb7u3
  SDP Session Name:       Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Legacy userfield parse: No
  Caller ID:              asterisk
  From: Domain:
  Record SIP history:     On
  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:  Enabled using externaddr
  Externhost:             <none>
  Externaddr:             110.142.242.138:0
  Externrefresh:          10
  Localnet:               10.61.0.0/255.255.192.0

Global Signalling Settings:
---------------------------
  Codecs:                 0x80000008050e (gsm|ulaw|alaw|g729|ilbc|h263|testlaw)
  Codec Order:            alaw:20,gsm:20,g729:20,ilbc:30
  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:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  1800 secs
  Outbound reg. timeout:  20 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
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                default
  Force rport:            No
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:
  MOH Interpret:          default
  MOH Suggest:
  Voice Mail Extension:   asterisk


> not able to make outbound call from Asterisk/1.8.13.1
> -----------------------------------------------------
>
>                 Key: ASTERISK-24762
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24762
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>         Environment:  == Using SIP RTP CoS mark 5
>     -- Executing [61452377795 at outgoing:1] NoOp("SIP/61500-00000018", "outgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795") in new stack
>     -- Auto fallthrough, channel 'SIP/61500-00000018' status is 'UNKNOWN'
>            Reporter: Ashish
>
> == Using SIP RTP CoS mark 5
>     -- Executing [61452377795 at outgoing:1] NoOp("SIP/61500-00000018", "outgoing CALLERID(all) "ASHISH" <61500> calling EXTEN 61452377795") in new stack
>     -- Auto fallthrough, channel 'SIP/61500-00000018' status is 'UNKNOWN'



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