Hello again, - I configured a 1:1 NAT to sipx and removed the old inbound NAT Rules - Using this neat little command (curl -s http://checkip.dyndns.org | sed 's/[a-zA-Z/<> :]//g') I confirmed that that the outbound NAT was working correctly... that is to say the externally detected IP matches the external IP configured in public DNS. (this brings up the question: If internal DNS uses the private LAN IP and external DNS uses the Public IP would that screw with me?) - I followed the steps outlined below concerning P-A-I - I applied the patch mentioned below.
The Polycom phones successfully place a call on hold but drop the call when attempting to resume the call. MOH is not currently set on the phone or the sipXbridge, and incoming calls are going to a hunt group (ext 300) Watching the CDR's the incoming number shows a call in progress to ext. 300 and when the call is placed on hold that call remains and whatever extension from the hunt group answered (i.e. 202) shows a call in progress to [email protected] That doesn't seem entirely correct to me??? Is it? Another bit of information I haven't added thus far but remembered might be important has to do with the configuration files generated by sipx for the Polycom Phones. Attempting to troubleshoot on my own I read this wiki page http://sipx-wiki.calivia.com/index.php/HowTo_configure_Polycom_SoundPoint_IP _phones_with_sipX It states "sipXconfig supports Polycom firmware releases 1.6.x and 2.x. By default newly created Polycom devices (or imported Polycom devices) are presumed to have 1.6 firmware installed. When adding or editing the phone you can tell sipXconfig which firmware version it uses (please not that 2.0 really means any 2.x firmare)." I ran the command to generate 2.x compatible files... Is this information outdated? Would running that command screw me up? The phones are currently running: Bootrom: 4.1.3 release Firmware: 3.1.3RevC Do I need to run 2.x firmware for compatibility? If so, does the bootrom version matter? Thanks for humoring me on this, Jonathan Petersen Ontra LLC www.ontraonline.com -----Original Message----- From: M. Ranganathan [mailto:[email protected]] Sent: Wednesday, July 22, 2009 8:02 AM To: Jonathan Petersen Cc: [email protected] Subject: Re: [sipx-users] FW: Small Business configuration suggestions 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_a nd_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/
