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. |