Hello
I trying to debug a strange problem, after two years of running (with very little problems) my asterisk box can not receive incoming calls but can make outgoing calls. The timing is strange as it stopped after FreePBX upgraded their servers. When I look at the asterisk CLI, I see that my system registers with the trunk and I receive an ACK from the trunk but when an outside call is made, I never see an invite hit the server. I have all the proper ports open (5060 and 10000-20000 UDP and TCP)… What am I missing, I have been trying to resolve this using their “PAID” support which was a huge waste of time (and money) so I am turning to the community to help me figure what can be going on. I also included a dump of the output when I run a sip set debug peer command on the trunk
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trixbox*CLI> sip set debug peer fpbx-1-192b5871
SIP Debugging Enabled for IP: 184.72.227.214:5060
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 184.72.227.214:5060:
REGISTER sip:trunk1.phonebooth.net SIP/2.0
Via: SIP/2.0/UDP 63.81.34.2:5060;branch=z9hG4bK4f29807f;rport
From: sip:[email protected];tag=as6c88351c
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 105 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“192b5871”, realm=“trunk.phonebooth.net”, algorithm=MD5, uri=“sip:trunk1.phonebooth.net”, nonce=“83718e6b-abee-49c6-9449-3539f8108efd”, response=“50033565dfbb9dc15391f02086a2159a”, qop=auth, cnonce=“79d37fee”, nc=00000003
Expires: 120
Contact: sip:[email protected]
Event: registration
Content-Length: 0
trixbox*CLI>
<— SIP read from 184.72.227.214:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.81.34.2:5060;received=63.81.34.2;branch=z9hG4bK4f29807f;rport=5060
From: sip:[email protected];tag=as6c88351c
To: sip:[email protected];tag=g4te76FDacH0S
Call-ID: [email protected]
CSeq: 105 REGISTER
Contact: sip:[email protected]:5060;expires=120
Date: Sat, 23 Jul 2011 17:35:26 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: precondition, path, replaces
Path: sip:pb2proxy-pro-aws03.phonebooth.net;lr
Content-Length: 0
<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)
Reliably Transmitting (NAT) to 184.72.227.214:5060:
OPTIONS sip:trunk1.phonebooth.net SIP/2.0
Via: SIP/2.0/UDP 63.81.34.2:5060;branch=z9hG4bK3eec3612;rport
From: “Unknown” sip:[email protected];tag=as05133286
To: sip:trunk1.phonebooth.net
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 23 Jul 2011 17:33:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
trixbox*CLI>
<— SIP read from 184.72.227.214:5060 —>
SIP/2.0 200 ok
Via: SIP/2.0/UDP 63.81.34.2:5060;branch=z9hG4bK3eec3612;rport=5060
From: “Unknown” sip:[email protected];tag=as05133286
To: sip:trunk1.phonebooth.net;tag=dae6a81a6d8d2d5d596768146f317b6f.078a
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: Phonebooth/1.0.0
Content-Length: 0