[asterisk-bugs] [JIRA] (ASTERISK-28595) Asterisk 15.7.2 with TLS
Akit vasava (JIRA)
noreply at issues.asterisk.org
Mon Oct 21 04:31:47 CDT 2019
Akit vasava created ASTERISK-28595:
--------------------------------------
Summary: Asterisk 15.7.2 with TLS
Key: ASTERISK-28595
URL: https://issues.asterisk.org/jira/browse/ASTERISK-28595
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: Channels/chan_pjsip
Affects Versions: 15.7.2
Environment: ubuntu 18
Reporter: Akit vasava
Severity: Critical
Hello All,
I have installed asterisk 15.7.2 version for TLS with twilio.(using pjsip(latest version)
For outgoing it is working fine from asterisk to twilio on 5061 port.
But for incoming calls , i have received call invite into asterisk server
but asterisk gives getting SIP/2.0 488 Not Acceptable Here
Below is the trunk and call information.
===========
Trunk info
=============
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/etc/asterisk/keys/asterisk.crt
priv_key_file=/etc/asterisk/keys/asterisk.key
local_net=10.176.120.0/16
external_media_address=20X.XXX.XXX.XX
external_signaling_address=20X.XXX.XXX.XX
method=tlsv1
verify_client=no
verify_server=no
allow_reload=no
tos=cs3
cos=3
[twilio-trunkstls](!)
type=endpoint
transport=transport-tls
media_encryption=sdes
media_encryption_optimistic=no
context=from-pstn
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
dtmf_mode=rfc2833
disallow=all
allow=ulaw
allow=alaw
[auth-out](!)
type=auth
auth_type=userpass
[twilio0tls](twilio-trunkstls)
aors=twilio0tls-aors
[twilio0tls-aors]
type=aor
contact=sip:haXXXXX-tls.pstn.twilio.com:5061
[twilio0tls-ident]
type=identify
endpoint=twilio0tls
match=54.172.60.0
match=54.172.60.1
match=54.172.60.2
match=54.172.60.3
============
incoming call invite
===============
<--- Received SIP request (1376 bytes) from TLS:54.172.60.3:41731 --->
INVITE sip:+12054306070 at 207.115.87.182:5061;transport=tls SIP/2.0
Record-Route: <sip:54.172.60.3:5061;transport=tls;r2=on;lr>
Record-Route: <sip:54.172.60.3:5060;r2=on;lr>
From: <sip:+919879491525 at hapoprivacy-tls.pstn.twilio.com:5060>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
To: <sip:+12054306070 at 207.115.87.182:5061;transport=tls>
CSeq: 468365 INVITE
Max-Forwards: 63
Diversion: <sip:+12054306070 at public-vip.us1.twilio.com>;reason=unconditional
Call-ID: 1139415727745a28bc3626fa32aba49c at 0.0.0.0
Via: SIP/2.0/TLS 54.172.60.3:5061;branch=z9hG4bK5223.247d3ad1.0
Via: SIP/2.0/UDP 172.18.46.138:5060;rport=5060;received=172.18.46.138;branch=z9hG4bK8744b1c2-cb34-42ca-af05-faf5019566c1_6772d868_460-8104884425645884892
Contact: <sip:+919879491525 at 172.18.46.138:5060;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,OPTIONS
User-Agent: Twilio Gateway
X-Twilio-AccountSid: AC96646d3bd96676097d567cfdada4e1d0
Content-Type: application/sdp
X-Twilio-CallSid: CA2da00e21afcc863e79fab1742356033c
Content-Length: 325
v=0
o=root 1977884913 1977884913 IN IP4 34.203.250.176
s=Twilio Media Gateway
c=IN IP4 34.203.250.176
t=0 0
m=audio 11216 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BUGkfXPcfjd254I+x7gkXkpz+Ob/cVz5ElgjisRS
== Setting global variable 'SIPDOMAIN' to '207.115.87.182'
<--- Transmitting SIP response (662 bytes) to TLS:54.172.60.3:41731 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 54.172.60.3:5061;rport=41731;received=54.172.60.3;branch=z9hG4bK5223.247d3ad1.0
Via: SIP/2.0/UDP 172.18.46.138:5060;rport=5060;received=172.18.46.138;branch=z9hG4bK8744b1c2-cb34-42ca-af05-faf5019566c1_6772d868_460-8104884425645884892
Record-Route: <sip:54.172.60.3:41731;transport=TLS;lr;r2=on>
Record-Route: <sip:54.172.60.3:5060;lr;r2=on>
Call-ID: 1139415727745a28bc3626fa32aba49c at 0.0.0.0
From: <sip:+919879491525 at hapoprivacy-tls.pstn.twilio.com>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
To: <sip:+12054306070 at 207.115.87.182>
CSeq: 468365 INVITE
Server: Asterisk PBX 15.7.2
Content-Length: 0
<--- Transmitting SIP response (716 bytes) to TLS:54.172.60.3:41731 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 54.172.60.3:5061;rport=41731;received=54.172.60.3;branch=z9hG4bK5223.247d3ad1.0
Via: SIP/2.0/UDP 172.18.46.138:5060;rport=5060;received=172.18.46.138;branch=z9hG4bK8744b1c2-cb34-42ca-af05-faf5019566c1_6772d868_460-8104884425645884892
Record-Route: <sip:54.172.60.3:41731;transport=TLS;lr;r2=on>
Record-Route: <sip:54.172.60.3:5060;lr;r2=on>
Call-ID: 1139415727745a28bc3626fa32aba49c at 0.0.0.0
From: <sip:+919879491525 at hapoprivacy-tls.pstn.twilio.com>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
To: <sip:+12054306070 at 207.115.87.182>;tag=50f6b727-4abd-4b61-80d7-ce735d3ba2c7
CSeq: 468365 INVITE
Server: Asterisk PBX 15.7.2
Content-Length: 0
<--- Received SIP request (463 bytes) from TLS:54.172.60.3:41731 --->
ACK sip:+12054306070 at 207.115.87.182:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 54.172.60.3:5061;branch=z9hG4bK5223.247d3ad1.0
From: <sip:+919879491525 at hapoprivacy-tls.pstn.twilio.com>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
Call-ID: 1139415727745a28bc3626fa32aba49c at 0.0.0.0
To: <sip:+12054306070 at 207.115.87.182>;tag=50f6b727-4abd-4b61-80d7-ce735d3ba2c7
CSeq: 468365 ACK
Max-Forwards: 70
User-Agent: Twilio Gateway
Content-Length: 0
Twilio team said that configuration looks fine from your side just raised the issues in asterisk forum.
Can you please help me out ?
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