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Summary:ASTERISK-19171: sip tcp fails with secret
Reporter:Sean Darcy (seandarcy)Labels:
Date Opened:2012-01-08 10:52:58.000-0600Date Closed:2018-01-02 08:44:26.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:1.8.9.0 10.1.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Fedora 15Attachments:
Description: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@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@10.10.11.180>;tag=as76f9dd12
To: <sip:145@officePBX>
Contact: <sip:office_incoming@10.10.11.180:5060;transport=TCP>
Call-ID: 214e3c9a2c002f986833365b38fb49b4@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@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@10.10.11.180>;tag=as76f9dd12
To: <sip:145@officePBX>;tag=as12632493
Call-ID: 214e3c9a2c002f986833365b38fb49b4@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
Comments:By: Joshua C. Colp (jcolp) 2017-12-19 06:21:04.011-0600

Are you still experiencing this problem with current chan_sip on a supported version of Asterisk?
By: Asterisk Team (asteriskteam) 2018-01-02 08:44:26.363-0600

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