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 203.166.6.160:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 203.166.6.160;branch=z9hG4bK0c27.74ffd7721a5646250d3561247ea31238.0;received=203.166.6.160
Via: SIP/2.0/UDP 203.166.6.160:5061;branch=z9hG4bK1cec556fe4fa80c8953aae50d0815b49;rport=5061
From: sip:0437885540@203.166.6.160;tag=579a9f9b0dbf23f8d044228292ad4270
To: sip:61386839094@203.166.6.160;tag=as28dd58ed
Call-ID: 553844DD-57C211DD-B750978D-74E33A45@202.85.241.127
CSeq: 200 INVITE
User-Agent: Dogtel PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
I cannot explain why incoming calls would want to connect to the same 203.166.6.160 (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.
extensions.conf:
[pennytel_1-out]
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
[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()
sip.conf:
[pennytel_1-out]
type=peer
secret=XXXXXX
username=61386839094
host=sip.pennytel.com
fromuser=61386839094
canrenvite=no
insecure=port,invite
nat=yes
exterip=123.2.141.176
context=pennytel_1-out
allow=all
register => 61386839094:xxx2@sip.pennytel.com/61386839094
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@10.88.33.240
of course there is no user s,
<------------>
asterisk01*CLI>
<— SIP read from 203.166.6.160:5060 —>
INVITE sip:s@10.88.33.240 SIP/2.0<===============================!!
Record-Route: sip:203.166.6.160;ftag=6af75ae234c77ff4c6517009a2dc8901;lr
Via: SIP/2.0/UDP 203.166.6.160;branch=z9hG4bK03c5.1cc050e20cd2a9d448acd02e50f6603a.0
Via: SIP/2.0/UDP 203.166.6.160:5061;branch=z9hG4bK0e8f93747db893fe446e9f1637922fa2;rport=5061
Max-Forwards: 16
From: sip:0387469386@203.166.6.160;tag=6af75ae234c77ff4c6517009a2dc8901
To: sip:61386839094@203.166.6.160
Call-ID: 8D038C35-597811DD-A070978D-74E33A45@202.85.241.127
CSeq: 200 INVITE
Contact: Anonymous sip:203.166.6.160:5061
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 203.166.6.160 : 5060 (no NAT)
Using INVITE request as basis request – 8D038C35-597811DD-A070978D-74E33A45@202.85.241.127
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 202.85.241.127:17774
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 202.85.241.127:17774
Looking for s in pennytel_1-out (domain 10.88.33.240)
=============================================================================
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 203.166.6.160 N 5060 Unmonitored
5004/5004 10.88.33.210 D 5060 Unmonitored
5010/5010 (Unspecified) D 0 Unmonitored
5002/5002 10.88.33.102 D 5060 Unmonitored
5000/5000 10.88.33.101 D 5060 Unmonitored
5001/5001 (Unspecified) D 0 Unmonitored
Pennytel/61386839094 203.166.6.160 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