[asterisk-bugs] [JIRA] (ASTERISK-24687) Asterisk behind NAT sets wrong Contact Header
Lukas Hauser (JIRA)
noreply at issues.asterisk.org
Tue Jan 27 10:09:34 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24687?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224640#comment-224640 ]
Lukas Hauser commented on ASTERISK-24687:
-----------------------------------------
Michael,
the server's LAN address of the asterisk 11.7 machine is 192.168.122.104 and the one of the asterisk 13.1 is 192.168.122.99.
They are both virtual machines using KVM and NAT with an own public IP for each server.
*sip settings asterisk 11.7*
{code}
asterisk11*CLI> sip show settings
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: Off
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm #Hostname#
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: Telephone Server
SDP Session Name: Telephone Server
SDP Owner Name: peter-marrat
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: Yes
T.38 EC mode: FEC
T.38 MaxDtgrm: 400
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: Yes
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: 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: public
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: asterisk
----
asterisk11*CLI>
{code}
*sip settings asterisk 13.1*
{code}
asterisk*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: 0.0.0.0:5060
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: No
Allow promisc. redir: No
Enable call counters: No
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: Telephone Server
SDP Session Name: Telephone Server
SDP Owner Name: peter-marrat
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: 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: (ulaw|alaw|gsm|h263)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
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: 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: asterisk
----
asterisk*CLI>
{code}
I've also tried it with the `realm` setting in 13.1, but there is no difference.
So far, I've not tried it with the latest version 11.
> 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.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!
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list