[asterisk-bugs] [JIRA] (ASTERISK-19050) Wrong transport for outgoing INVITE

Asterisk Team (JIRA) noreply at issues.asterisk.org
Tue Jan 2 08:45:48 CST 2018


     [ https://issues.asterisk.org/jira/browse/ASTERISK-19050?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team closed ASTERISK-19050.
------------------------------------

    Resolution: Suspended

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

> Wrong transport for outgoing INVITE
> -----------------------------------
>
>                 Key: ASTERISK-19050
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-19050
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Messaging, Channels/chan_sip/TCP-TLS
>    Affects Versions: 1.8.7.2
>         Environment: Minimal CentOS 5.7, Asterisk 1.8.7.0 from repo, updated to 1.8.7.2, behind the NAT.
>            Reporter: Dmitry Alekseev
>            Assignee: Joshua Colp
>
> 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 at 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 at multifon.ru>;tag=as176e1f32
> To: <sip:7AAAAAAAAAA at multifon.ru>
> Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as176e1f32
> To: <sip:7AAAAAAAAAA at multifon.ru>;tag=2CB932463135364126DE1600
> Call-ID: 20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1
> CSeq: 105 REGISTER
> Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060;transport=TCP>;expires=130
> Content-Length: 0
> Service-Route: <sip:7AAAAAAAAAA at 193.201.229.35:5060;transport=tcp;lr>
> <------------->
> --- (9 headers 0 lines) ---
> Scheduling destruction of SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 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 at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>
> Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at 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 at multifon.ru>;tag=as653efae6
> To: <sip:7BBBBBBBBBB at multifon.ru>;tag=aprqngfrt-k04r2m20000c6
> Contact: <sip:7AAAAAAAAAA at 10.128.1.225:5060>
> Call-ID: 21bd2e403656248b19d83565196334ea at 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 at multifon.ru>;tag=as653efae6'
> localhost*CLI> 
> Really destroying SIP dialog '21bd2e403656248b19d83565196334ea at multifon.ru' Method: INVITE
> localhost*CLI> 
> Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: OPTIONS
> localhost*CLI> 
> Really destroying SIP dialog '20861aac7e7569e62c3608ff7d62cd9e at 127.0.0.1' Method: REGISTER
> localhost*CLI> 
>  << Hangup on console >> 
> localhost*CLI> 
> {noformat}



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