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ID | Category | Severity | Reproducibility | Date Submitted | Last Update | |||||||
0000390 | [Yate - Yet Another Telephony Engine] engine | major | always | 2015-02-16 09:39 | 2015-02-22 13:49 | |||||||
Reporter | schorpp | View Status | public | |||||||||
Assigned To | ||||||||||||
Priority | normal | Resolution | open | |||||||||
Status | new | Product Version | SVN | |||||||||
Summary | 0000390: SIP TLS signalling failure with provider dus.net, dus.net CRYPTO-ENGINE 2.2 -> SIP/488 | |||||||||||
Description |
SIP TLS signalling failure with provider dus.net dus.net CRYPTO-ENGINE 2.2 -> SIP/488 Maybe it needs other ciphers or G.726 codecs offered, Twinkle offers G.726 but Yate does not in INVITE. Maybe 2 acounts in accfile using proxy.dus.net trigger a colliding bug? [dus_net] enabled=yes protocol=sip username=**********434 description=**********434 interval=600 ;authname=metoo password=************** domain=proxy.dus.net registrar=proxy.dus.net ;outbound=proxy.dus.net localaddress=yes keepalive=5000 ip_transport=tls [dus_fax] enabled=yes protocol=sip username=**********384 description=**********384 interval=600 ;authname=metoo password=**************** domain=proxy.dus.net registrar=proxy.dus.net ;outbound=proxy.dus.net localaddress=yes keepalive=5000 ;ip_transport=tls I'll recheck the dus.net docs. |
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Additional Information |
2015-02-16_08:12:45.241539 <sip:INFO> 'udp:0.0.0.0:5060' received 900 bytes SIP message from 127.0.0.1:5085 [0xeb7620] ------ INVITE sip:**********@localhost SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5085;rport;branch=z9hG4bKghfdzkxc Max-Forwards: 70 To: <sip:**********@localhost> From: "210" <sip:210@localhost>;tag=hlxqr Call-ID: jxnqgcbynxelzbd@tom3.fritz.box CSeq: 388 INVITE Contact: <sip:210_localhost@127.0.0.1:5085> Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.4.2 Content-Length: 396 v=0 o=twinkle 980190720 69060975 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 8000 RTP/AVP 8 0 3 102 104 103 105 98 97 99 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:102 G726-16/8000 a=rtpmap:104 G726-32/8000 a=rtpmap:103 G726-24/8000 a=rtpmap:105 G726-40/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=ptime:30 ------ ----------------------------------------- ------ 2015-02-16_08:12:45.321509 <sip:INFO> 'tls:**********:55057-83.125.8.71:5061' sending 'INVITE sip:**********@proxy.dus.net' 0x7fe0ec03cd80 [0xec76e0] ------ INVITE sip:**********@proxy.dus.net SIP/2.0 Max-Forwards: 19 Via: SIP/2.0/TLS **********:55057;alias;rport;branch=z9hG4bK1394119314 From: "210" <sip:**********434@localhost>;tag=1877497748 To: <sip:**********@proxy.dus.net> Call-ID: 1498547685@localhost User-Agent: YATE/5.4.1 Contact: <sip:**********434@**********:55057> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO CSeq: 27 INVITE Proxy-Authorization: Digest username="**********434", realm="dus.net", nonce="***", uri="sip:**********@proxy.dus.net", response="****", algorithm=MD5 Content-Type: application/sdp Content-Length: 430 v=0 o=yate 1424070765 1424070765 IN IP4 ********** s=SIP Call c=IN IP4 ********** t=0 0 m=audio 24418 RTP/AVP 8 0 3 103 102 104 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:103 SPEEX/16000 a=rtpmap:102 SPEEX/8000 a=rtpmap:104 SPEEX/32000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:72ixzHZDTymLirzksut7YmdYCVouZua2hK6VeOou a=encryption:optional ------ 2015-02-16_08:12:45.419280 <sip:INFO> 'tls:**********:55057-83.125.8.71:5061' received 349 bytes SIP message [0xec76e0] ------ SIP/2.0 100 Call will be processed now... Via: SIP/2.0/TLS **********:55057;alias;rport=55057;branch=z9hG4bK1394119314;received=********** From: "210" <sip:**********434@localhost>;tag=1877497748 To: <sip:**********@proxy.dus.net> Call-ID: 1498547685@localhost CSeq: 27 INVITE Server: DUS.NET TERM-SIP-ENGINE 2.1 Content-Length: 0 ------ 2015-02-16_08:12:45.429928 <sip:INFO> 'tls:**********:55057-83.125.8.71:5061' received 358 bytes SIP message [0xec76e0] ------ SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TLS **********:55057;alias;rport=55057;branch=z9hG4bK1394119314;received=********** From: "210" <sip:**********434@localhost>;tag=1877497748 To: <sip:**********@proxy.dus.net> Call-ID: 1498547685@localhost CSeq: 27 INVITE Server: DUS.NET TERM-SIP-ENGINE 2.1 Content-Length: 0 ------ 2015-02-16_08:12:45.461070 <sip:INFO> 'tls:**********:55057-83.125.8.71:5061' received 456 bytes SIP message [0xec76e0] ------ SIP/2.0 488 Not acceptable here Via: SIP/2.0/TLS **********:55057;received=**********;alias;rport=55057;branch=z9hG4bK1394119314 From: "210" <sip:**********434@localhost>;tag=1877497748 To: <sip:**********@proxy.dus.net>;tag=as0aee5495 Call-ID: 1498547685@localhost CSeq: 27 INVITE Server: dus.net CRYPTO-ENGINE 2.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 ------ |
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Tags | No tags attached. | |||||||||||
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(0000578) schorpp (reporter) 2015-02-16 10:46 |
Maybe the User-Agent: YATE/5.4.1 is only accepted for business accounts, CW is not detected, trying to fake User-Agent next. |
(0000579) schorpp (reporter) 2015-02-16 10:55 |
No, nothing such mentioned here: https://www.dus.net/en/rates/ratecomparechart [^] |
(0000580) schorpp (reporter) 2015-02-18 12:23 |
http://docs.yate.ro/wiki/SIP_in_Yate [^] outdated: http://yate.null.ro/archive/?action=show_msg&actionargs[]=71&actionargs[]=32 [^] grep shows no source code for codec ITU G.726 in yate. CLOSED: WONTFIX: Provider enforces patented codecs - avoid. |
(0000582) schorpp (reporter) 2015-02-22 11:58 |
RESOLVED: FIXED by using yate SIP SBC Mode with forward_sdp=enable in ysipchan.conf , ${rtp_forward}possible=;rtp_forward=yes in regexroute.conf. |
(0000586) schorpp (reporter) 2015-02-22 13:49 edited on: 2015-02-22 13:51 |
REVERT: BREAKS OTHER ...providers than dus.net. Needs exclusive setup: ^n\(.*\)$=line/\1;line=dus_net;caller=000*********;${rtp_forward}possible=;rtp_forward=yes Incoming calls untested for provider dus.net. |
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