Hi All,
Just a follow up. I deployed the fixed version of siproxd into an 15
phone environment last night and all seems to be working perfectly.
Thanks Jonathan for the idea of a OpenSER, I will have to take a look.
If these solutions require any tweaking of the phone configs then they
will not
be viable in this situation. The voIP provider does not allow any
config changes.
Note about PFsense Support:
This is the second instance where I had an issue that I all my
exhaustive attempts failed to produce a fixed solution. The first time
I was under heavy pressure to get an network up and this last time
with getting these phones to work. I can't express enough the prompt
response and professionalism that the guys at Centipede have provided.
Not only do they respond and actually look into the problem quickly,
they did this on a Sunday! if you don't have a support contract I
highly recommend that you do.
Cheers!
Lee
On Jun 16, 2008, at 3:52 PM, Jonathan GF wrote:
Althought the problem seems to be solver i would propose Lee to use
a SIP router like OpenSER and/or a Media Gateway like Asterisk for
VoIP purposes.
The only equipments that will cross the firewall will be those ones
and Lee can have the network infrastructure as he needs, among the
benefits to have a VoIP (why not rather say signaling system) in the
network.
Regards,
Jonathan GF
Lee J. Imber wrote:
Hi All,
I am stuck and hoping someone here can help.
Here is the situation.
I have 10 SIP phones Polycom IP320's on a internal 10.0.0.x net.
These
phones then get dhcp from the pfsense 1.2-RELEASE box. Then out a
cable modem to the phone provider.
The Problem.
I can only get one phone to work. The first phone that boots works,
then remaining phones don't. When I say they don't work, they boot
fine, get Ip information but I get no dial tone and I cannot make
inbound or outbound calls. The phone that boot first works perfectly.
I have tried all the various NAT tweaks I can think of like enabling
static port and AON, nothing works same issue.
I read :
"SIP Limitation - By default, all TCP and UDP traffic other than SIP
and IPsec gets the source port rewritten. More information on this
can
be found in the static port documentation. Because this source port
rewriting is how pf tracks which internal IP made the connection to
the given external server, and most all SIP traffic uses the same
source port, only one SIP device can connect simultaneously to a
single server on the Internet. Unless your SIP devices can operate
with source port rewriting (most can't), you cannot use multiple
phones with a single outside server without using a dedicated public
IP per device. The sipproxd package will provide a work around for
this issue, and is currently under development."
OK, forget playing with rules/nat.
I have installed siproxd and been digging through that documentation
and testing with no luck.
This is where I am, anyone have a working siproxd.conf that would be
similar to my topology?
Any pointers?
Thanks,
Lee
