Hi All
Ok. The problem was with sip station. There was a problem with there sip proxy. They did something on there side and my incoming calls started to work.
Here is a layout of events and solutions. I;m using freepbx with asterisk 1.6 from rentpbx.com
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When I first created my account with sip station the admin panel in sip station was blank it was missing the sip trunk username, password and gateway info. I emailed sip station support and after some running around they resolved the problem.
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I had problems with outgoing calls in the beginning. So i deleted all my configuration and used the Auto-configure string for freepbx provided by sip station inside the admin panel on sip station.com to configure my asterisk system. The string created two sip trunks. The trunks worked for going calls no problem.
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Once the trunks were created I had problems with incoming calls. When calling my DID the message would say this “Number is not in Use” and hang up.
I ran a sip debug commands:
rasterisk | tee ast-deb-incoming.txt
core set debug 9
sip set debug on
!netstat -aunp --> include the exclamation
Once I ran the debug I saw the following error message from my asterisk box. I also stopped my firewall using this command “service iptables stop” works with centos before running th debug to rule out any port firewall issues.
Sip station sip proxy had a problem with my sip registration request.
SIP/2.0 401 not authorized
Via: SIP/2.0/UDP 173.231.00.00:5060;branch=z9hG4bK39599686
From: sip:[email protected];tag=as2d0d0a89
To: sip:[email protected];tag=dae6a81a6d8d2d5d596768146f317b6f.c5
44
Call-ID: [email protected]
CSeq: 106 REGISTER
Server: Phonebooth/1.0.0
Content-Length: 0
This is what my asterisk sent to sip station sip proxy for a registration request. The output above is the response from sip station.
REGISTER sip:trunk1.phonebooth.net SIP/2.0
Via: SIP/2.0/UDP 173.231.00.00:5060;branch=z9hG4bK39599686
Max-Forwards: 70
From: sip:[email protected];tag=as2d0d0a89
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 106 REGISTER
User-Agent: FPBX-2.7.0(1.6.2.15)
Expires: 120
Contact: sip:[email protected]
Content-Length: 0
I emailed this sip debug info to sip station email suport and they resolved the problem. There email support is not that good they take awhile to respond and sometimes you have to email them twice. Also start a new email when communicating with them do not reply. Also turn your firwall back on once complete. “service iptables start”
In any case my system is up and running. Problem resolved.
Also here is my sip station trunk configuration. This worked for me.
Trunk Description: fpbx-1234
Outbound Caller ID: Anything you want
Outgoing Settings:
Trunk Name: fpbx-1234
PEER Details
disallow=all
allow=ulaw&g729
context=from-pstn
type=peer
insecure=very
qualify=yes
sendrpid=yes
trustrpid=yes
dtmfmode=rfc2833
username=your username
secret=your password
host=trunk1.phonebooth.net
Incoming Settings:
USER Context: incoming
host=trunk1.phonebooth.net
type=friend
insecure=port,invite
dissallow=all
allow=ulaw
context=from-trunk
Registration:
yourusername:[email protected]