Hi there.
I have about 300 SIP extensions (yealink T22) with TLS and SRTP enabled registered to asterisk 11.17.1 and everything seems to works as it should for few weeks.
BUT For a reason a can’t put my finger on the extensions with TLS and SRTP stop working while the others using udp still do. After that whenever a secure extension tries to start a call I see the following log lines:
[Jun 5 11:27:28] VERBOSE[28837][C-0001dfcb] netsock2.c: == Using SIP VIDEO TOS bits 136
[Jun 5 11:27:28] VERBOSE[28837][C-0001dfcb] netsock2.c: == Using SIP VIDEO CoS mark 6
[Jun 5 11:27:28] VERBOSE[28837][C-0001dfcb] netsock2.c: == Using SIP RTP TOS bits 184
[Jun 5 11:27:28] VERBOSE[28837][C-0001dfcb] netsock2.c: == Using SIP RTP CoS mark 5
[Jun 5 11:27:28] WARNING[28837][C-0001dfcb] chan_sip.c: Rejecting secure audio stream without encryption details: audio 11796 RTP/SAVP 0 8 18 101
A core restart fixes the problem until it happens again (It happened 2 times already) and saddly I couldn’t get debug on both ocasions.
Any thought on how to fix this?
Thankx!
SIP settings:
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: aa.bb.cc.dd:5061
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
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 11.17.1
SDP Session Name: Asterisk PBX 11.17.1
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: Unknown
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 : 1200000 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: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|h263p|h264)
Codec Order: alaw:20,ulaw:20,gsm:20,h263p:0,h264:0
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 240
RTP Hold Timeout: 600
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: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Refuse
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: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language: pt_BR
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97