A little more information would be helpful. What carrier are you using for 
siptrunking? The issues you are mentioning suggest a trunking issue rather than 
firewall. A lot of ITSP's do not support every feature.
You mentioned port 5800, normally that would be 5080. Is 5800 the signalling 
port your ITSP is contacting you on? Are you allowing all traffic out via your 
firewall? Do you have MOH turned on for both the handsets and the ITSP (I would 
test with the handsets off and ITSP via sipXbridge on)?

>>> "Jonathan Petersen" <[email protected]> 07/21/09 6:00 PM >>>
Hello, 

 

I am attempting to use sipx as a solution for a small business telephone
system.  I have 10 Polycom IP330's and one Polycom IP650.  

 

I would like to have the following configuration:

- Firewall (either m0n0wall or PFSense) on a DSL circuit

- SIP Trunk using sipxbridge

- Support the following

            - Single Incoming number rings all handsets

            - Blind and consultative transfers

            - Hold (preferably with MOH)

            - MWI

            - Buddy Lists

            - Conferencing

 

I have incoming and outgoing calls working successfully but whenever I
attempt to transfer a call or place somebody on hold the call disconnects.
I am assuming that I am having either a NAT problem or a configuration error
in the phone group / or line configuration.  I have played around with both
a hunt group and a shared line configuration as a solution for ringing all
the incoming phones at the same time and encounter call hold and transfer
issues with both setups.

 

Here is an overview of the current setup:

 

Dedicated DSL Circuit (Connection tested at
http://myvoipspeed.visualware.com/ MOS score of 4.0 with .4ms jitter and 0%
packet loss)

 

Firewall m0n0wall

Inbound NAT and Firewall Rules

-          TCP/UDP 5800 from WAN IP to sipx internal IP

-          TCP/UDP 30000 - 31000 from WAN IP to sipx internal IP

 

Sipx from installer CD 4.0.1

-          Single server running park running SIP Trunking, Conferencing,
Management, Primary SIP Router, and Voicemail Server Roles

-          Followed directions here
http://sipx-wiki.calivia.com/index.php/SipXbridge_Overview_and_Configuration
(multiple times)

-          Sipx is not the DNS server but the DNS configuration test
completes just fine

-          Sipx is the DHCP server

-          Sipx is managing the phones

 

Polycom Phones

Bootrom: 4.1.3 release

Firmware: 3.1.3RevC

 

Any help would be greatly appreciated and upon a reliable working setup I'd
be glad to publish a very specific tutorial fit for this use case.

 

Thanks,

 

Jonathan Petersen

 

 

Ontra LLC

www.ontraonline.com

 

</[email protected]>
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