Tony,
Thanks for the quick reply! The 5800 port was a typo (sorry) it is indeed
5080 (checked and double checked firewall and sipx config). The ITSP is a
company called Binary Telecom (www.binarytelecom.com). They are a local
company and I have direct contact with the owner (thought that might be
beneficial). All outbound traffic is enabled from the subnet sipx is on to
the internet. I have tried MOH both ways (ITSP off/handset on, ITSP
on/handset off). I recalled reading something about having both of them on
being a problem so didn't try that.
Is there something specific I should be looking for on the sip trunk? I
asked if they supported the following as mentioned in the SIP Trunking
howto:
Following are the minimal requirements for interoperability :
* Must support RFC 3261
* Must support Re-INVITE for mid-call codec renegotiation.
* Must support Session Timer.
* Must support P-Asserted-Identity for call forwarding.
They do not support P-Asserted-Identity, so I read into it and thought that
would only affect call forwarding to external numbers (I only need to
transfer calls to internal extensions so didn't figure this would be a
problem??? Can you clarify?).
Once again thank you for your reply,
Jonathan Petersen
Ontra LLC
www.ontraonline.com
-----Original Message-----
From: Tony Graziano [mailto:[email protected]]
Sent: Tuesday, July 21, 2009 3:16 PM
To: [email protected]; [email protected]
Subject: Re: [sipx-users] Small Business configuration suggestions
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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