sean darcy
2012-Jan-02 04:34 UTC
[asterisk-users] tcp version of toronto - osaka doesn't work
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 at 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 Is the book just out of date?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
sean darcy
2012-Jan-02 16:21 UTC
[asterisk-users] tcp version of toronto - osaka doesn't work
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 at 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)