[asterisk-bugs] [JIRA] (ASTERISK-24855) NOTIFY with sips
Slava Bendersky (JIRA)
noreply at issues.asterisk.org
Thu Mar 26 19:03:34 CDT 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24855?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225637#comment-225637 ]
Slava Bendersky commented on ASTERISK-24855:
--------------------------------------------
Hello Rusty,
Here sip settings for this server.
canlpbx01*CLI> sip show settings
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: 0.0.0.0:5060
TLS SIP Bindaddress: 0.0.0.0:5061
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
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 mydomain.conf
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: FPBX-2.11.0(11.15.1)
SDP Session Name: Asterisk PBX 11.15.1
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: capbxsrv01
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: Yes
T.38 EC mode: Redundancy
T.38 MaxDtgrm: 415
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: 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: Yes
Jitterbuffer forced: No
Jitterbuffer max size: 200
Jitterbuffer resync: 1000
Jitterbuffer impl: fixed
Jitterbuffer log: Yes
Network Settings:
---------------------------
SIP address remapping: Enabled using externaddr
Externhost: <none>
Externaddr: mypubip:0
Externrefresh: 10
Localnet: mylansubnet/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: (ulaw|g729|h264)
Codec Order: ulaw:20,g729:20,h264:0
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 10
RTP Timeout: 30
RTP Hold Timeout: 300
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: 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: Yes
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: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
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: *97
> NOTIFY with sips
> ----------------
>
> Key: ASTERISK-24855
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24855
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/Subscriptions, Channels/chan_sip/TCP-TLS
> Affects Versions: 11.15.1, 12.8.1
> Environment: Linux canlpbx02. 2.6.32-431.el6.x86_64
> Reporter: Slava Bendersky
> Assignee: Rusty Newton
> Severity: Critical
>
> NOTIFY with TLS send wrong protocol schema in routing headers.
> sip:sips
> {noformat}
> ----------------------------------------
> -- 2015-03-08 20:30:19 - Received from ASTERISK_PUB_IP:5061 at 192.168.88.254:5069
> NOTIFY sips:10102 at 192.168.88.254:5069 SIP/2.0
> Via: SIP/2.0/TLS ASTERISK_PUB_IP:5061;branch=z9hG4bK5a08186f;rport
> Max-Forwards: 70
> Route: <sips:10102 at 192.168.88.254:5069>
> From: "capbxsrv01" <sip:capbxsrv01 at ASTERISK_PUB_IP>;tag=as6f8f9fea
> To: <sip:sips:10102 at 192.168.88.254:5069>;tag=d92d0bcee3
> Contact: <sip:capbxsrv01 at ASTERISK_PUB_IP:5061;transport=TLS>
> Call-ID: 35667ff0c4c88d84
> CSeq: 102 NOTIFY
> User-Agent: Asterisk PBX(11.15.1)
> Event: message-summary
> Content-Type: application/simple-message-summary
> Subscription-State: active
> Content-Length: 105
> Messages-Waiting: yes
> Message-Account: sip:*97 at ASTERISK_PUB_IP;transport=TLS
> Voice-Message: 24/4 (0/0)
> ----------------------------------------
> -- 2015-03-08 20:30:19 - Sent to ASTERISK_PUB_IP:5061 from 192.168.88.254:5069
> SIP/2.0 200 OK
> Via: SIP/2.0/TLS ASTERISK_PUB_IP:5061;branch=z9hG4bK5a08186f;rport=5061;received=ASTERISK_PUB_IP
> From: "capbxsrv01" <sip:capbxsrv01 at ASTERISK_PUB_IP>;tag=as6f8f9fea
> To: <sip:sips:10102 at 192.168.88.254:5069>;tag=d92d0bcee3
> Call-ID: 35667ff0c4c88d84
> CSeq: 102 NOTIFY
> Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE
> Server: Media5-fone/4.1.6.3283 Android/5.0.1
> Supported: eventlist, replaces, timer
> Content-Length: 0
> {noformat}
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