[asterisk-bugs] [JIRA] (ASTERISK-28595) Asterisk 15.7.2 with TLS
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Mon Oct 21 04:31:47 CDT 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-28595?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248476#comment-248476 ]
Asterisk Team commented on ASTERISK-28595:
------------------------------------------
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> 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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