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

[sean darcy]

On 01/02/2012 11:30 AM, sean darcy wrote:
> 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 Is the book just out of date?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
>

OK, the book is out of date. Do Not put the name of the local 
device/user in the register statement.

sean