Summary: | ASTERISK-29004: SIP/2.0 488 Not Acceptable Here when configured with PJSIP/TLS | ||
Reporter: | Jack Brain (razormouse) | Labels: | |
Date Opened: | 2020-07-22 08:18:15 | Date Closed: | 2020-07-22 08:18:18 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | pjproject/pjsip |
Versions: | 17.6.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | Operating System: Kali GNU/Linux Rolling Kernel: Linux 4.4.0-186-generic Architecture: x86-64 Asterisk: 17.6.0 | Attachments: | |
Description: | - We have build Asterisk 17.6.0 and PJSIP from the source on a VPS
- We configured TLS, generated client and server certificate - Registered the Blink client for testing, success - While making a call from Blink_1 to Blink_2 or vice-versa, it fails with an error {code} <--- Received SIP request (1021 bytes) from TLS:43.249.37.23:54700 ---> INVITE sip:2222@94.140.114.51 SIP/2.0 Via: SIP/2.0/TLS 192.168.75.143:49485;rport;branch=z9hG4bKPj28ae9d1c90c443fbac7f6021e4771d5c;alias Max-Forwards: 70 From: "1111" <sip:1111@94.140.114.51>;tag=54c6492c375e44bf8a3599fe8047016f To: <sip:2222@94.140.114.51> Contact: <sip:80631759@192.168.75.143:49453;transport=tls> Call-ID: 73fa29d60a5a4ea7989d2d175ea91342 CSeq: 16687 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink 3.2.0 (Windows) Content-Type: application/sdp Content-Length: 429 v=0 o=- 3804383127 3804383127 IN IP4 192.168.75.143 s=Blink 3.2.0 (Windows) t=0 0 m=audio 50048 RTP/AVP 113 9 0 8 101 c=IN IP4 192.168.75.143 a=rtcp:50049 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=zrtp-hash:1.10 6171448735d90ecf08cf6a7fefedf5369e0c4b0825c13d3b30741d5182bdff7f a=sendrecv <--- Transmitting SIP response (566 bytes) to TLS:43.249.37.23:54700 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.75.143:49485;rport=54700;received=43.249.37.23;branch=z9hG4bKPj28ae9d1c90c443fbac7f6021e4771d5c;alias Call-ID: 73fa29d60a5a4ea7989d2d175ea91342 From: "1111" <sip:1111@94.140.114.51>;tag=54c6492c375e44bf8a3599fe8047016f To: <sip:2222@94.140.114.51>;tag=z9hG4bKPj28ae9d1c90c443fbac7f6021e4771d5c CSeq: 16687 INVITE WWW-Authenticate: Digest realm="asterisk",nonce="1595419528/95038c7ada5d1c4091cb7649c149cd06",opaque="5a2d21b867e7b7d7",algorithm=md5,qop="auth" Server: Asterisk PBX 17.6.0 Content-Length: 0 <--- Received SIP request (423 bytes) from TLS:43.249.37.23:54700 ---> ACK sip:2222@94.140.114.51 SIP/2.0 Via: SIP/2.0/TLS 192.168.75.143:49485;rport;branch=z9hG4bKPj28ae9d1c90c443fbac7f6021e4771d5c;alias Max-Forwards: 70 From: "1111" <sip:1111@94.140.114.51>;tag=54c6492c375e44bf8a3599fe8047016f To: <sip:2222@94.140.114.51>;tag=z9hG4bKPj28ae9d1c90c443fbac7f6021e4771d5c Call-ID: 73fa29d60a5a4ea7989d2d175ea91342 CSeq: 16687 ACK User-Agent: Blink 3.2.0 (Windows) Content-Length: 0 <--- Received SIP request (1314 bytes) from TLS:43.249.37.23:54700 ---> INVITE sip:2222@94.140.114.51 SIP/2.0 Via: SIP/2.0/TLS 192.168.75.143:49485;rport;branch=z9hG4bKPjbf7e99f726d74c4ebbbb508d7033dea9;alias Max-Forwards: 70 From: "1111" <sip:1111@94.140.114.51>;tag=54c6492c375e44bf8a3599fe8047016f To: <sip:2222@94.140.114.51> Contact: <sip:80631759@192.168.75.143:49453;transport=tls> Call-ID: 73fa29d60a5a4ea7989d2d175ea91342 CSeq: 16688 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink 3.2.0 (Windows) Authorization: Digest username="1111", realm="asterisk", nonce="1595419528/95038c7ada5d1c4091cb7649c149cd06", uri="sip:2222@94.140.114.51", response="7b6971ff7f3d73d8cb888640056d6e3e", algorithm=md5, cnonce="5bacef95a6f243e19c4cfc0b41ebb5c9", opaque="5a2d21b867e7b7d7", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 429 v=0 o=- 3804383127 3804383127 IN IP4 192.168.75.143 s=Blink 3.2.0 (Windows) t=0 0 m=audio 50048 RTP/AVP 113 9 0 8 101 c=IN IP4 192.168.75.143 a=rtcp:50049 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=zrtp-hash:1.10 6171448735d90ecf08cf6a7fefedf5369e0c4b0825c13d3b30741d5182bdff7f a=sendrecv == Setting global variable 'SIPDOMAIN' to '94.140.114.51' <--- Transmitting SIP response (368 bytes) to TLS:43.249.37.23:54700 ---> SIP/2.0 100 Trying Via: SIP/2.0/TLS 192.168.75.143:49485;rport=54700;received=43.249.37.23;branch=z9hG4bKPjbf7e99f726d74c4ebbbb508d7033dea9;alias Call-ID: 73fa29d60a5a4ea7989d2d175ea91342 From: "1111" <sip:1111@94.140.114.51>;tag=54c6492c375e44bf8a3599fe8047016f To: <sip:2222@94.140.114.51> CSeq: 16688 INVITE Server: Asterisk PBX 17.6.0 Content-Length: 0 <--- Transmitting SIP response (422 bytes) to TLS:43.249.37.23:54700 ---> SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/TLS 192.168.75.143:49485;rport=54700;received=43.249.37.23;branch=z9hG4bKPjbf7e99f726d74c4ebbbb508d7033dea9;alias Call-ID: 73fa29d60a5a4ea7989d2d175ea91342 From: "1111" <sip:1111@94.140.114.51>;tag=54c6492c375e44bf8a3599fe8047016f To: <sip:2222@94.140.114.51>;tag=a7cdfce3-d8be-42b1-b0db-47e437493be3 CSeq: 16688 INVITE Server: Asterisk PBX 17.6.0 Content-Length: 0 <--- Received SIP request (418 bytes) from TLS:43.249.37.23:54700 ---> ACK sip:2222@94.140.114.51 SIP/2.0 Via: SIP/2.0/TLS 192.168.75.143:49485;rport;branch=z9hG4bKPjbf7e99f726d74c4ebbbb508d7033dea9;alias Max-Forwards: 70 From: "1111" <sip:1111@94.140.114.51>;tag=54c6492c375e44bf8a3599fe8047016f To: <sip:2222@94.140.114.51>;tag=a7cdfce3-d8be-42b1-b0db-47e437493be3 Call-ID: 73fa29d60a5a4ea7989d2d175ea91342 CSeq: 16688 ACK User-Agent: Blink 3.2.0 (Windows) Content-Length: 0 {code} - pjsip.conf: {code} [default] type=transport protocol=tls bind=94.140.114.51:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [1111] type=aor max_contacts=1 remove_existing=yes [1111] type=auth auth_type=userpass username=1111 password=1111 [1111] type=endpoint aors=1111 auth=1111 context=default disallow=all allow=GSM allow=ulaw allow=g726 allow=g729 allow=speex allow=g722 allow=iLBC dtmf_mode=rfc4733 media_encryption=sdes [2222] type=aor max_contacts=1 remove_existing=yes [2222] type=auth auth_type=userpass username=2222 password=2222 [2222] type=endpoint aors=2222 auth=2222 context=default disallow=all allow=GSM allow=ulaw allow=g726 allow=g729 allow=speex allow=g722 allow=iLBC dtmf_mode=rfc4733 media_encryption=sdes [3333] type=aor max_contacts=1 remove_existing=yes [3333] type=auth auth_type=userpass username=3333 password=3333 [3333] type=endpoint aors=3333 auth=3333 context=default disallow=all allow=gsm allow=ulaw allow=g726 allow=g729 dtmf_mode=rfc4733 media_encryption=sdes {code} - extensions.conf {code} [general] static=yes writeprotect=no priorityjumping=no autofallthrough=yes clearglobalvars=no ;[local] ;exten=>1111,1,Dial(PJSIP/1111,20) ;exten=>2222,1,Dial(PJSIP/2222,20) ;exten=>3333,1,Dial(PJSIP/3333,20) [default] exten => 1111,1,Dial(PJSIP/1111,20) exten => 2222,1,Dial(PJSIP/2222,20) exten => 3333,1,Dial(PJSIP/3333,20) {code} | ||
Comments: | By: Asterisk Team (asteriskteam) 2020-07-22 08:18:18.030-0500 We appreciate the difficulties you are facing, however information request type issues 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. If this issue is actually a bug please use the Bug issue type instead. 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 By: Asterisk Team (asteriskteam) 2020-07-22 08:18:18.701-0500 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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. |