Sip 404 not found pennytel

I am trying to set up asterisk, to use SIP provider Pennytel

I can register no problem, and make outgoing calls.

I can see incoming calls hit the asterisk box:

<— Reliably Transmitting (NAT) to —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP;branch=z9hG4bK0c27.74ffd7721a5646250d3561247ea31238.0;received=
Via: SIP/2.0/UDP;branch=z9hG4bK1cec556fe4fa80c8953aae50d0815b49;rport=5061
From: sip:0437885540@;tag=579a9f9b0dbf23f8d044228292ad4270
To: sip:61386839094@;tag=as28dd58ed
Call-ID: 553844DD-57C211DD-B750978D-74E33A45@
CSeq: 200 INVITE
User-Agent: Dogtel PBX
Supported: replaces
Content-Length: 0

I cannot explain why incoming calls would want to connect to the same (via and from the the SIP header are the same) which is the IP address of the pennytel proxy. no where in the SIP debug output do I see the IP address of my asterisk box. The 61386839094 number is my DID number @ pennytel. yes I have forwarded UDP 5060 on my router.


exten => _003XXXXXXXX,1,Dial(SIP/${EXTEN:1}@pennytel_1-out,30)
exten => _004xxxxxxxx,1,Dial(SIP/${EXTEN:1}@pennytel_1-out,30)
exten => 61386839094,2,Dial(SIP/5000,25,Ttr)
include = dogtel_internal

exten => 5000,1,Dial(SIP/5000,15)
exten => 5000,2,Voicemail(5000@voicemail_dogtel)
exten => 5000,3,playback(vm-goodbye)
exten => 5000,3,Hangup()



register =>

Furthermore, when I dial 61386839094 from the asterisk console the 5000 phone rings
any ideas?

actually, i just found out, the calls come in as an INVITE to s@

of course there is no user s,

<— SIP read from —>
INVITE sip:s@ SIP/2.0<===============================!!
Record-Route: sip:;ftag=6af75ae234c77ff4c6517009a2dc8901;lr
Via: SIP/2.0/UDP;branch=z9hG4bK03c5.1cc050e20cd2a9d448acd02e50f6603a.0
Via: SIP/2.0/UDP;branch=z9hG4bK0e8f93747db893fe446e9f1637922fa2;rport=5061
Max-Forwards: 16
From: sip:0387469386@;tag=6af75ae234c77ff4c6517009a2dc8901
To: sip:61386839094@
Call-ID: 8D038C35-597811DD-A070978D-74E33A45@
CSeq: 200 INVITE
Contact: Anonymous sip:
Expires: 300
User-Agent: Sippy
cisco-GUID: 2365660053-1501041117-2469134363-1395158146
h323-conf-id: 2365660053-1501041117-2469134363-1395158146
Content-disposition: session
Content-Length: 366
Content-Type: application/sdp


— (17 headers 16 lines) —
Sending to : 5060 (no NAT)
Using INVITE request as basis request – 8D038C35-597811DD-A070978D-74E33A45@
Found peer 'pennytel_1-out’
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port
Found description format G729 for ID 18
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format G726-32 for ID 98
Found description format GSM for ID 3
Found description format telephone-event for ID 101
Found description format CN for ID 19
Capabilities: us – 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer – audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined – 0x90f (g723|gsm|ulaw|alaw|g726|g729)
Non-codec capabilities (dtmf): us – 0x1 (telephone-event), peer – 0x3 (telephone-event|CN), combined – 0x1 (telephone-event)
Peer audio RTP is at port
Looking for s in pennytel_1-out (domain


furthermore is see 2 peers registered with pennytel:

asterisk01*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
pennytel_1-out/6138683909 N 5060 Unmonitored
5004/5004 D 5060 Unmonitored
5010/5010 (Unspecified) D 0 Unmonitored
5002/5002 D 5060 Unmonitored
5000/5000 D 5060 Unmonitored
5001/5001 (Unspecified) D 0 Unmonitored
Pennytel/61386839094 5060 Unmonitored

I have no idea where that Pennytel registered user comes from as it is not defined in sip.conf.

your help is greatly appreciated.

remember: not all clever people work for cisco