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Summary:ASTERISK-28595: Asterisk 15.7.2 with TLS
Reporter:Akit vasava (ankitvasava09)Labels:
Date Opened:2019-10-21 04:30:21Date Closed:2019-10-21 06:01:19
Priority:CriticalRegression?
Status:Closed/CompleteComponents:Channels/chan_pjsip
Versions:15.7.2 Frequency of
Occurrence
Related
Issues:
Environment:ubuntu 18Attachments:
Description: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@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@hapoprivacy-tls.pstn.twilio.com:5060>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
To: <sip:+12054306070@207.115.87.182:5061;transport=tls>
CSeq: 468365 INVITE
Max-Forwards: 63
Diversion: <sip:+12054306070@public-vip.us1.twilio.com>;reason=unconditional
Call-ID: 1139415727745a28bc3626fa32aba49c@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@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@0.0.0.0
From: <sip:+919879491525@hapoprivacy-tls.pstn.twilio.com>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
To: <sip:+12054306070@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@0.0.0.0
From: <sip:+919879491525@hapoprivacy-tls.pstn.twilio.com>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
To: <sip:+12054306070@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@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@hapoprivacy-tls.pstn.twilio.com>;tag=78849536_6772d868_8744b1c2-cb34-42ca-af05-faf5019566c1
Call-ID: 1139415727745a28bc3626fa32aba49c@0.0.0.0
To: <sip:+12054306070@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 ?


Comments:By: Asterisk Team (asteriskteam) 2019-10-21 04:30:22.458-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
By: Joshua C. Colp (jcolp) 2019-10-21 06:01:11.045-0500

We appreciate the difficulties you are facing, however this does not appear to be a bug report and your request or comments would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines