[asterisk-bugs] [JIRA] (ASTERISK-19171) sip tcp fails with secret

Joshua Colp (JIRA) noreply at issues.asterisk.org
Tue Dec 19 06:21:08 CST 2017


     [ https://issues.asterisk.org/jira/browse/ASTERISK-19171?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-19171:
-----------------------------------

    Assignee: Sean Darcy
      Status: Waiting for Feedback  (was: Open)

Are you still experiencing this problem with current chan_sip on a supported version of Asterisk?

> sip tcp fails with secret
> -------------------------
>
>                 Key: ASTERISK-19171
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-19171
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.9.0, 10.1.0
>         Environment: Fedora 15
>            Reporter: Sean Darcy
>            Assignee: Sean Darcy
>
> tcp sip fails if "secret" is used.
> I'm trying to set up sip over tcp between home (10.1.0-rc1) and office (1.8.9.0-rc1). This works if I don't have "secret" in "office_incoming", but fails if I do.
> On home ( I've tried both "secret" and "remotesecret"):
> [home-outgoing]
> type=friend
> transport=tcp
> ;;remotesecret=office
> secret=office
> fromuser=office_incoming
> host=officePBX
> disallow=all
> allow=g722
> allow=ulaw
> allow=g729
> on office (I've also tried type "friend"):
> [office_incoming]
> type=user
> transport=tcp
> secret=office
> context=longdistance
> disallow=all
> allow=g722
> allow=ulaw
> allow=gsm
> on office:
> sip show users
> office_incoming            office                            longdistance     No   Yes
> sip debug on office:
> INVITE sip:145 at officePBX SIP/2.0
> Via: SIP/2.0/TCP 10.10.11.180:5060;branch=z9hG4bK3fa92fe8;rport
> Max-Forwards: 70
> From: "Home" <sip:office_incoming at 10.10.11.180>;tag=as76f9dd12
> To: <sip:145 at officePBX>
> Contact: <sip:office_incoming at 10.10.11.180:5060;transport=TCP>
> Call-ID: 214e3c9a2c002f986833365b38fb49b4 at 10.10.11.180:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 10.1.0-rc1
> Date: Sun, 08 Jan 2012 16:25:03 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 312
> v=0
> o=root 512097222 512097222 IN IP4 <homeIPaddr>
> s=Asterisk PBX 10.1.0-rc1
> c=IN IP4 <homeIPaddr>
> t=0 0
> m=audio 14308 RTP/AVP 0 9 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> <------------->
> --- (14 headers 14 lines) ---
>   == Using UDPTL TOS bits 184
>   == Using UDPTL CoS mark 5
> Sending to <homeIPaddr>:60142 (NAT)
> Using INVITE request as basis request - 214e3c9a2c002f986833365b38fb49b4 at 10.10.11.180:5060
> Found peer 'office_incoming' for 'office_incoming' from <homeIPaddr>:60142
> <--- Reliably Transmitting (NAT) to <homeIPaddr>:60142 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/TCP 10.10.11.180:5060;branch=z9hG4bK3fa92fe8;received=<homeIPaddr>;rport=60142
> From: "Home" <sip:office_incoming at 10.10.11.180>;tag=as76f9dd12
> To: <sip:145 at officePBX>;tag=as12632493
> Call-ID: 214e3c9a2c002f986833365b38fb49b4 at 10.10.11.180:5060
> CSeq: 102 INVITE
> Server: Asterisk PBX 1.8.9.0-rc1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="531eb693"
> Content-Length: 0



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