Summary: | ASTERISK-19050: Wrong transport for outgoing INVITE | ||
Reporter: | Dmitry Alekseev (rara avis) | Labels: | |
Date Opened: | 2011-12-15 04:35:58.000-0600 | Date Closed: | 2018-01-02 08:44:24.000-0600 |
Priority: | Major | Regression? | Yes |
Status: | Closed/Complete | Components: | Channels/chan_sip/Messaging Channels/chan_sip/TCP-TLS |
Versions: | 1.8.7.2 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Minimal CentOS 5.7, Asterisk 1.8.7.0 from repo, updated to 1.8.7.2, behind the NAT. | Attachments: | |
Description: | INVITE is sent via UDP despite previous REGISTER via TCP and TCP being declared as primary outboud transport.
REGISTERing to external provider via TCP - Ok. But outgoing INVITE to the same provider is sent via UDP, despite TCP being configured primary outgoing transport, 403 Forbidden as result. All combinations of "transport=tcp" or "transport=tcp,udp" in general or peer section or in both - behave the same. 1.8.7.0 - the same. 1.6.2.21 - working as expected (INVITE via TCP) sip.conf: {noformat} [general] context=default allowguest=no allowoverlap=no alwaysauthreject=yes udpbindaddr = 0.0.0.0 tcpenable = yes tcpbindaddr = 0.0.0.0 transport=tcp,udp srvlookup=yes nat = yes disallow=all allow=alaw allow=g729 allow=g723 allow=ulaw dtmfmode=inband canreinvite=no externip=AAA.BBB.CCC.DDD register = tcp://7AAAAAAAAAA@multifon.ru:password:7AAAAAAAAAA@sbc.megafon.ru/7AAAAAAAAAA [2provider] type=peer host = multifon.ru username = 7AAAAAAAAAA secret = password context = default outboundproxy = sbc.megafon.ru fromdomain = multifon.ru fromuser = 7AAAAAAAAAA authuser = 7AAAAAAAAAA insecure = port,invite {noformat} extensions.conf: {noformat} [general] static = yes writeprotect = no clearglobalvars = no [default] exten=>_7XXXXXXXXXX,1,Dial(SIP/${EXTEN}@2provider,30,r) {noformat} show's: {noformat} localhost*CLI> core show version Asterisk 1.8.7.2 built by root @ localhost.localdomain on a i686 running Linux on 2011-12-09 17:52:28 UTC localhost*CLI> localhost*CLI> sip show settings Global Settings: ---------------- UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: 0.0.0.0:5060 TLS SIP Bindaddress: Disabled Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: No Allow subscriptions: Yes Allow overlap dialing: No Allow promisc. redir: No Enable call counters: No SIP domain support: No Realm. auth: No Our auth realm asterisk Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk PBX 1.8.7.2 SDP Session Name: Asterisk PBX 1.8.7.2 SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Legacy userfield parse: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 60000 ms Q.850 Reason header: No Store SIP_CAUSE: No Network QoS Settings: --------------------------- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: No Network Settings: --------------------------- SIP address remapping: Disabled, no localnet list Externhost: <none> Externaddr: AAA.BBB.CCC.DDD:0 Externrefresh: 10 Global Signalling Settings: --------------------------- Codecs: 0x10d (g723|ulaw|alaw|g729) Codec Order: alaw:20,g729:20,g723:30,ulaw:20 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: Yes Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: <not set> Session Timers: Accept Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 500 Timer T1 minimum: 100 Timer B: 32000 No premature media: Yes Max forwards: 70 Default Settings: ----------------- Allowed transports: TCP,UDP Outbound transport: TCP Context: default Force rport: Yes DTMF: inband Qualify: 0 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk ---- localhost*CLI> sip show peer 2provider * Name : 2provider Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 7AAAAAAAAAA FromDomain : multifon.ru Port 5060 Callgroup : Pickupgroup : MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : port,invite Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No Outb. proxy : sbc.megafon.ru DTMFmode : inband Timer T1 : 500 Timer B : 32000 ToHost : multifon.ru Addr->IP : 193.201.229.35:5060 Defaddr->IP : (null) Prim.Transp. : TCP Allowed.Trsp : TCP,UDP Def. Username: 7AAAAAAAAAA SIP Options : (none) Codecs : 0x10d (g723|ulaw|alaw|g729) Codec Order : (alaw:20,g729:20,g723:30,ulaw:20) Auto-Framing : No 100 on REG : No Status : Unmonitored Useragent : Reg. Contact : Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No localhost*CLI> {noformat} debug: {noformat} localhost*CLI> sip set debug peer 2provider localhost*CLI> SIP Debugging Enabled for IP: 193.201.229.35 localhost*CLI> [Dec 14 16:20:19] NOTICE[3535]: chan_sip.c:12596 sip_reregister: -- Re-registration for 7AAAAAAAAAA@sbc.megafon.ru localhost*CLI> REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to 193.201.229.35:5060: REGISTER sip:multifon.ru SIP/2.0 Via: SIP/2.0/TCP 10.128.1.225:5060;branch=z9hG4bK3cedc749;rport Max-Forwards: 70 From: <sip:7AAAAAAAAAA@multifon.ru>;tag=as176e1f32 To: <sip:7AAAAAAAAAA@multifon.ru> Call-ID: 20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1 CSeq: 105 REGISTER User-Agent: Asterisk PBX 1.8.7.2 Authorization: Digest username="7AAAAAAAAAA", realm="BREDBAND", algorithm=MD5, uri="sip:multifon.ru", nonce="MTMyMzgzNjIzNDpfLjfk775Bd2kMNs4bxIr3", response="2fdc731e5f472d43843edc9d95b8ea9e", opaque="MTMyMzgzNjIzNDpfLjfk775Bd2kMNs4bxIr3", qop=auth, cnonce="74cc256c", nc=00000003 Expires: 120 Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060;transport=TCP> Content-Length: 0 --- localhost*CLI> <--- SIP read from TCP:193.201.229.35:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK3cedc749;rport=36591 From: <sip:7AAAAAAAAAA@multifon.ru>;tag=as176e1f32 To: <sip:7AAAAAAAAAA@multifon.ru>;tag=2CB932463135364126DE1600 Call-ID: 20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1 CSeq: 105 REGISTER Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060;transport=TCP>;expires=130 Content-Length: 0 Service-Route: <sip:7AAAAAAAAAA@193.201.229.35:5060;transport=tcp;lr> <-------------> --- (9 headers 0 lines) --- Scheduling destruction of SIP dialog '20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1' in 32000 ms (Method: REGISTER) [Dec 14 16:20:19] NOTICE[3547]: chan_sip.c:20155 handle_response_register: Outbound Registration: Expiry for sbc.megafon.ru is 130 sec (Scheduling reregistration in 115 s) localhost*CLI> console dial 7BBBBBBBBBB localhost*CLI> Audio is at 5060 localhost*CLI> Adding codec 0x8 (alaw) to SDP localhost*CLI> Adding codec 0x4 (ulaw) to SDP localhost*CLI> Reliably Transmitting (NAT) to 193.201.229.35:5060: INVITE sip:7BBBBBBBBBB@multifon.ru SIP/2.0 Via: SIP/2.0/UDP 10.128.1.225:5060;branch=z9hG4bK2d2b39d7;rport Max-Forwards: 70 From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6 To: <sip:7BBBBBBBBBB@multifon.ru> Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060> Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.7.2 Date: Wed, 14 Dec 2011 10:20:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 202 v=0 o=root 518834463 518834463 IN IP4 10.128.1.225 s=Asterisk PBX 1.8.7.2 c=IN IP4 10.128.1.225 t=0 0 m=audio 16650 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- localhost*CLI> <--- SIP read from UDP:193.201.229.35:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK2d2b39d7;rport=1027 From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6 To: <sip:7BBBBBBBBBB@multifon.ru> Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru CSeq: 102 INVITE <-------------> --- (6 headers 0 lines) --- <--- SIP read from UDP:193.201.229.35:5060 ---> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.128.1.225:5060;received=AAA.BBB.CCC.DDD;branch=z9hG4bK2d2b39d7;rport=1027 From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6 To: <sip:7BBBBBBBBBB@multifon.ru>;tag=aprqngfrt-k04r2m20000c6 Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru CSeq: 102 INVITE Reason: Q.850;cause=55;text="Call Terminated" <-------------> --- (7 headers 0 lines) --- Transmitting (NAT) to 193.201.229.35:5060: ACK sip:7BBBBBBBBBB@multifon.ru SIP/2.0 Via: SIP/2.0/UDP 10.128.1.225:5060;branch=z9hG4bK2d2b39d7;rport Max-Forwards: 70 From: "asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6 To: <sip:7BBBBBBBBBB@multifon.ru>;tag=aprqngfrt-k04r2m20000c6 Contact: <sip:7AAAAAAAAAA@10.128.1.225:5060> Call-ID: 21bd2e403656248b19d83565196334ea@multifon.ru CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.7.2 Content-Length: 0 --- localhost*CLI> [Dec 14 16:20:43] WARNING[3535]: chan_sip.c:19680 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:7AAAAAAAAAA@multifon.ru>;tag=as653efae6' localhost*CLI> Really destroying SIP dialog '21bd2e403656248b19d83565196334ea@multifon.ru' Method: INVITE localhost*CLI> Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1' Method: OPTIONS localhost*CLI> Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e@127.0.0.1' Method: REGISTER localhost*CLI> << Hangup on console >> localhost*CLI> {noformat} | ||
Comments: | By: Dmitry Alekseev (rara avis) 2011-12-16 00:15:05.421-0600 Workaround: outboundproxy = tcp://... By: Luke H (luckman212) 2012-01-05 10:49:29.821-0600 Is this fixed as of the 1.8.8.1 or 1.8.9.0-rc1 ? Just curious as I may be having an issue related to this. By: Dmitry Alekseev (rara avis) 2012-01-05 12:39:48.294-0600 The problem still persist in 1.8.8.1 . By: Joshua C. Colp (jcolp) 2017-12-19 06:24:35.930-0600 Are you still experiencing this in chan_sip as of a currently supported version of Asterisk? By: Asterisk Team (asteriskteam) 2018-01-02 08:44:24.369-0600 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |