Hi, FreeSWITCH gurus! I
need your help!
First off, I am new to FS and I am new to Internet telephony
as well. Heck, I am new to the concept of NAT, UPnP, etc.,
so please bear with my ignorance.
I subscribe to a VoIP service at home, with which I get one
DID. They supply me a VoIP adapter. Their expected usage is
for you to plug in analog phones to the analog phone jacks
in the VoIP adapter. However, It also has four Ethernet LAN
ports and it acts as a router. You can also access it from
the LAN side and register with its built-in SIP gateway.
What I would like to do is to run FS (Windows version) on
one of the Windows PCs, have it register with the SIP
gateway, and have it act as an AA or IVR. For testing, I am
having it just play music.
When I tried the same idea with AsterikWin32, it worked just
as I had hoped; it answered incoming calls automatically.
However, I somehow cannot make it work with FS. I simulate
incoming calls to my DID number with Skype's Sypeout. It
fails after a short while with such error messages as
"network error." It appears the call was never answered.
FS is assigned an extension number 7 at the gateway. When I
call extension 7 from a different extension (at the gateway
level, not an extension inside FS), FS does answer the call
and I hear music. FS fails to answer only incoming calls
from outside.
I think my FS configuration is fairly standard. I created an
external SIP profile for the gateway under
conf/sip_profiles/external/ and modified
conf/dialplan/public/00_inbound_did.xml so incoming calls to
the gateway will be transferred to an extension within FS.
FS's messages and logs, plus the result of packet captures
indicate that FS
thinks it has answered the call,
and goes on to initiate media communication. I see RTP
packets going from FS to the SIP gateway. What's different
from AsteriskWin32's case is that there are no RTP packets
coming back from the SIP gateway to FS. Turning of the
firewall of the PC does not seem to change the result in any
way.
For your perusal, I have created the following logs of
communication between FS/AsteriskWin32 and the SIP gateway:
- AsteriskWin32's case
- FreeSWITCh's case
The IP address of the SIP gateway is
192.168.11.250,
and that of the PC FS/AsteriskWin32 resides in is
192.168.11.11.
My DID number is masked as ABCDEFGHIJ. I do not know if it
gives you any useful information, but those files include
the registration phase.
FS's log was taken at a
different time, so it does not entirely match the
packet captures.
I also have the corresponding Pcap files. Please let me know
if you need them.
I am not entirely sure, but I think as far as what I'd like
to do is concerned, NAT is not going to be an issue, because
the FS/AsteriskWin32 PC and the SIP gateway (its LAN side IP
address) are on the same subnet (192.168.11/24). At this
time, I do not need to access FS from the Internet.
Finally, I will give you more details about my setup, which
may or may not be relevant to this issue.
My home LAN is set up this way:
http://i139.photobucket.com/albums/q302/starplatina/For%20Blogs/HomeLANSetup.jpg
Please note that there are
two layers of NAT, and
that in the inner layer, two NAT devices exist. I know it
looks convoluted, but there are logical reasons for this
setup.
The VoIP service provider only supports the PCMU codec. The
music file I prepared for this testing is encoded in PCMU,
so codecs will not be an issue.
Please do not hesitate to ask if you have any questions.
Thanks for your help in advance!
Yasuro