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

[sean darcy]

On 01/02/2012 11:21 AM, sean darcy wrote:
> 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 Is the book just out of date?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)
>
>

As I think about it, isn't this a problem with 10.1.0 on toronto.

The INVITE is correct:

INVITE sip:osaka@<toronto>:5060;transport=TCP SIP/2.0

so why isn't 10.1.0 looking for peer "osaka"?

Is it simply a mistake that it's taking the user from the FROM header 
rather than the INVITE?

sean