[asterisk-users] tcp version of toronto - osaka doesn't work

[sean darcy]

On 01/01/2012 11:34 PM, sean darcy wrote:
> I'm trying to setup a simple tcp sip connection based on the toronto
> osaka example in the Asterisk book.
>
> On the remote box (osaka) (1.8.9.0-rc1):
>
> [toronto]
> type=friend
> transport=tcp
> secret=welcome
> context=toronto_incoming
> host=dynamic
> disallow=all
> allow=ulaw
>
> sip show peer toronto
>
>
> * Name : toronto
> Secret : <Set>
> MD5Secret : <Not set>
> Remote Secret: <Not set>
> Context : toronto_incoming
> ........
> Useragent : Asterisk PBX 10.1.0-rc1
> Reg. Contact : sip:osaka@<toronto>:5060;transport=TCP
>
>
> On the home box (toronto) (10.1.0-rc1):
>
> register => tcp://toronto:welcome@officePBX/osaka
> [osaka]
> type=friend
> transport=tcp
> secret=welcome
> context=incoming
> host=dynamic
> disallow=all
> allow=ulaw
>
> But make a call from the remote Dial(SIP/toronto) , and the home cli shows:
>
> Call from '' (<remote>:5060) to extension 'osaka' rejected because
> extension not found in context 'default'.
>
> which makes no sense to me at all. Doesn't the string after the "/" in
> register refer to the user/device on the box doing the register? Doesn't
> it tell the device on the remote host which local device to connect to?
> i.e., toronto at remote > osaka at home ?? And where's context "default"
> coming from?
>
> Is the book just out of date? Or is tcp not ready?
>
> sean
>

Looks like tcp is messed up. Or is my setup somehow flawed?  Does anyone 
have tcp working?

Turning on sip debug on toronto gave the below INVITE. Notice From: 
"Anonymous" <sip:Anonymous at anonymous.invalid>

Why isn't this toronto <sip:toronto@<osaka>> ?  As it is, Anonymous 
becomes the peer/user, which is not found. Then osaka is viewed as the 
extension - not the peer - and context default is searched for osaka.

<--- SIP read from TCP:<osaka>:5060 --->
INVITE sip:osaka@<toronto>:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP <osaka>:5060;branch=z9hG4bK41111f7e;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous at anonymous.invalid>;tag=as697266a6
To: <sip:osaka@<toronto>:5060;transport=TCP>
Contact: <sip:Anonymous at 184.75.103.142:5060;transport=TCP>
Call-ID: 6f7df020162fa79f7e58b2015ab0f410@<osaka>:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.9.0-rc1
Date: Mon, 02 Jan 2012 15:58:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 1399746571 1399746571 IN IP4 <osaka>
s=Asterisk PBX 1.8.9.0-rc1
c=IN IP4 <osaka>
t=0 0
m=audio 11112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
   == Using UDPTL TOS bits 184
   == Using UDPTL CoS mark 5
Sending to <osaka>:5060 (NAT)
Using INVITE request as basis request - 
6f7df020162fa79f7e58b2015ab0f410@<osaka>:5060
No matching peer for 'Anonymous' from '<osaka>:5060'
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|speex|g722), peer - 
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <osaka>:11112
Looking for osaka in default (domain <toronto>)

<--- Reliably Transmitting (NAT) to <osaka>:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 
<osaka>:5060;branch=z9hG4bK41111f7e;received=<osaka>;rport=5060
From: "Anonymous" <sip:Anonymous at anonymous.invalid>;tag=as697266a6
To: <sip:osaka@<toronto>:5060;transport=TCP>;tag=as3e025900
Call-ID: 6f7df020162fa79f7e58b2015ab0f410 at 184.75.103.142:5060
CSeq: 102 INVITE
Server: Asterisk PBX 10.1.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Jan  2 10:58:22] NOTICE[6432]: chan_sip.c:23063 handle_request_invite: 
Call from '' (<osaka>:5060) to extension 'osaka' rejected because 
extension not found in context 'default'.
Scheduling destruction of SIP dialog 
'6f7df020162fa79f7e58b2015ab0f410@<osaka>:5060' in 32000 ms (Method: INVITE)