ASTERISK-24855: TLS NOTIFY with SIPS uses "sip:sips" in To header

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Summary: ASTERISK-24855: TLS NOTIFY with SIPS uses "sip:sips" in To header

Reporter: Slava Bendersky (volga629) Labels:

Date Opened: 2015-03-09 06:11:07 Date Closed:

Priority: Critical Regression?

Status: Open/New Components: Channels/chan_sip/Subscriptions Channels/chan_sip/TCP-TLS

Versions: 11.15.1 12.8.1 13.18.4 Frequency of
Occurrence Constant

Related
Issues:

Environment: Linux canlpbx02. 2.6.32-431.el6.x86_64 Attachments: ( 0) client_side_debug.txt
( 1) server_side_debug.txt

Description: 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@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@192.168.88.254:5069>
From: "capbxsrv01" <sip:capbxsrv01@ASTERISK_PUB_IP>;tag=as6f8f9fea
To: <sip:sips:10102@192.168.88.254:5069>;tag=d92d0bcee3
Contact: <sip:capbxsrv01@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@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@ASTERISK_PUB_IP>;tag=as6f8f9fea
To: <sip:sips:10102@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}

Comments: By: Rusty Newton (rnewton) 2015-03-12 08:49:32.312-0500

Thanks for the report Slava. Please attach your sanitized full sip.conf configuration as well.

Use "Send Back" or "Enter Feedback" once you have the configuration attached. Thanks.
By: Slava Bendersky (volga629) 2015-03-26 19:02:12.634-0500

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
By: Slava Bendersky (volga629) 2015-03-31 21:58:17.374-0500

Rusty,
I can comment in ticket, but I don't see where options for

Use "Send Back" or "Enter Feedback" once you have the configuration attached.
By: Richard Mudgett (rmudgett) 2015-03-31 22:13:16.980-0500

Those buttons show up when the issue status is "Waiting for feedback". When the issue is in that state it doesn't show up with his filters as an issue needing attention.
By: Rusty Newton (rnewton) 2015-04-08 16:51:54.291-0500

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

By: Rusty Newton (rnewton) 2015-04-08 16:54:06.040-0500

[~volga629] The SIP clients I used to attempt reproduction apparently don't support the SIPS URI scheme and apparently the Media5-fone you are using requires commercial components to support TLS/SIPS.

Can you provide additional debug as noted above?

We'd like to see a complete subscription and call, interleaved with the verbose and debug logger channels. Thanks!
By: Rusty Newton (rnewton) 2015-04-08 16:54:50.375-0500

Also, please note if the "sip:sips:" issue in the To header is the only problem you found. I saw that you said "headers".
By: Slava Bendersky (volga629) 2015-04-10 08:38:27.836-0500

NOTIFY debugs
By: Slava Bendersky (volga629) 2015-04-10 08:38:33.281-0500

We using Media5-fone, because I found it most stable one for android devices and TLS function in general works good.
Please see attached files.
1. From asterisk side.
2. From client side ( Media5-fone have built in tracer for sip)

All those debug indicate to problems that after first SUBSCRIBE asterisk generate NOTIFY with sip:sips
By: Rusty Newton (rnewton) 2015-04-10 09:55:57.108-0500

Re-attaching Slava's debug with txt extension so it'll be easier to access for everyone.
By: Rusty Newton (rnewton) 2015-04-10 09:58:50.103-0500

[~volga629] can you attach the debug as requested - following this guide: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

*I don't see any logging channels other than your SIP trace which means you didn't follow the guide as requested.*

We need a full debug log, including your SIP trace with warning, error, notice, verbose, debug type log channels. Please turn verbose and debug up to the levels indicated in the guide.

Thanks!
By: Rusty Newton (rnewton) 2015-04-27 10:38:01.630-0500

I'm opening the issue, but we could still use the debug requested.