[asterisk-bugs] [JIRA] (ASTERISK-24687) Asterisk behind NAT sets wrong Contact Header
Lukas Hauser (JIRA)
noreply at issues.asterisk.org
Tue Jan 27 11:29:34 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24687?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224643#comment-224643 ]
Lukas Hauser commented on ASTERISK-24687:
-----------------------------------------
I've now tried it even with the *asterisk 11.15.0* version (LAN address 192.168.122.122).
*Same issue here!!*
{code}
sterisk11-latest*CLI> sip show settings
Global Settings:
----------------
UDP Bindaddress: [::]:5060
** Additional Info:
[::] may include IPv4 in addition to IPv6, if such a feature is enabled in the OS.
TCP SIP Bindaddress: [::]:5060
TLS SIP Bindaddress: Disabled
Videosupport: No
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: 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 11.15.0
SDP Session Name: Asterisk PBX 11.15.0
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
Call Events: 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: 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: #ServersPublicIP#:5060
Externrefresh: 10
Localnet: 192.168.122.0/255.255.255.255
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|h263|testlaw)
Codec Order: none
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: 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 403:0
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: unauthenticated
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
----
{code}
sip show peer
{code}
asterisk11-latest*CLI> sip show peer 799
* Name : 799
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : localsets-common
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language :
Tonezone : <Not set>
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 3347
Insecure : no
Force rport : Auto (Yes)
Symmetric RTP: No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 4294967295
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
TrustIDOutbnd: Legacy
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : auto
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : #ServerPublicIP#:2048
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 799
SIP Options : 100rel from-change replaces replace timer
Codecs : (ulaw|alaw|g722)
Codec Order : (g722:20,alaw:20,ulaw:20)
Auto-Framing : No
Status : OK (55 ms)
Useragent : snom300/8.7.3.25.5
Reg. Contact : sip:799 at 192.168.1.41:2048;line=sotxm5mm
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
{code}
Now we get the same retransmit error, with a way different output: (full version attached above)
{code}
[Jan 27 18:22:18] DEBUG[1446] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #175
{code}
> Asterisk behind NAT sets wrong Contact Header
> ---------------------------------------------
>
> Key: ASTERISK-24687
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24687
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 13.1.0
> Environment: ubuntu 14.04.1
> Reporter: Lukas Hauser
> Assignee: Lukas Hauser
> Severity: Critical
> Attachments: debug-asterisk11.15.log, debug.log, deubg-asterisk11.log, sip-settings.log
>
>
> If asterisk is behind a statically configured NAT (e.g. with iptables), the externaddr or localnet option does not work.
> The Contact Header still contains the private IP address.
> The media_address option works.
> It also works as defined with version 11.7.0~dfsg-1ubuntu1.
> The Contact Header does not get updated in version 13.1 and 13.1.0-rc2.
> Therefore, I guess it is a bug in 13?
> Thanks!
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