On Tue, Jul 21, 2009 at 6:34 PM, Jonathan Petersen<[email protected]> wrote: > 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.
Please read wiki page update on the implications of not supporting P-A-I and read how to configure account without P-A-I here: http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration#6_Configure_an_ITSP_account_that_is_managed_by_SipXbridge Please make sure you apply the following patch before the test: http://track.sipfoundry.org/secure/attachment/20447/patch4.zip > > 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]> > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- M. Ranganathan _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
