Hello, everyone, again.
TLS port of my installation in 58790, not 5061.
I have a problem and it will drive me crazy.
When user from internet calling to my FreePBX everything goes right, untill user B pick up phone.
Then FreePBX send 200, OK with IP and port 5061 (!!!).
Softphone from other side replies Ack to IP:5061. Nothing happens! Of course my port is 58790!!!
PBX retransmitting few times 200 OK and then disconnecting the call with cause No user responding, 18.
My PBX configuration:
sip_general_additional.conf:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
vmexten=*97
useragent=FPBX-14.0.1.4(14.6.0)
disallow=all
allow=alaw
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
media_address=x.x.x.x
alwaysauthreject=yes
rtpend=20000
context=clients_route
rtpstart=10000
tcpenable=yes
callevents=no
tlsprivatekey=/etc/asterisk/keys/asterisk.key
tlscertfile=/etc/asterisk/keys/asterisk.crt
bindport=5060
jbenable=no
checkmwi=10
maxexpiry=3600
minexpiry=60
srvlookup=no
tlsenable=yes
allowguest=no
notifyhold=yes
rtptimeout=30
canreinvite=no
tlsbindaddr=0.0.0.0:58790
rtpkeepalive=0
videosupport=no
defaultexpiry=120
notifyringing=yes
maxcallbitrate=384
rtpholdtimeout=300
g726nonstandard=no
registertimeout=20
tlsclientmethod=sslv2
registerattempts=0
tlsdontverifyserver=yes
nat=force_rport,comedia
ALLOW_SIP_ANON=no
callerid=Unknown
externip=x.x.x.x
localnet=10.10.10.0/24
language=en
sip_general_custom.conf:
externaddr=x.x.x.x
nat=force_rport
localnet=10.10.10.0/255.255.255.0
user configuration:
[33333]
deny=0.0.0.0/0.0.0.0
secret=12345
dtmfmode=rfc2833
canreinvite=no
context=clients_route
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=force_rport,comedia
port=5060
qualify=yes
qualifyfreq=120
transport=tls,udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/33333
permit=0.0.0.0/0.0.0.0
callerid=Carl <33333>
callcounter=yes
faxdetect=no
sip show settings
server*CLI> sip show settings
Global Settings:
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: 0.0.0.0:58790
RTP 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: 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: FPBX-14.0.1.4(14.6.0)
SDP Session Name: Asterisk PBX 14.6.0
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: Unknown
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: 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: Enabled using externaddr
Externhost:
Externaddr: x.x.x.x:0
Externrefresh: 10
Localnet: 10.10.10.0/255.255.255.0
10.10.10.0/255.255.255.0
Global Signalling Settings:
Codecs: (alaw)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
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:No
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:
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: clients_route
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Language: ru
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
RTCP Multiplexing: No
I tried all combinations of NAT parameters. Nothing helped.
I dont know where FreePBX sending port 5061, tls making troubleshooting real difficult.
When I changing TLS port to 5061 - no problem! everything work fine!
Maybe anyone faced with same issue?