Based on that DNS document, my assumption is that I should have a local sipX box at each location rather than a cluster residing in the datacenter? Polycom's are on 3.2.2, I will downgrade them to 3.1.3c and see what happens. All phones are showing up registered. What I mean by SIP inspection is that the firewall is essentially handling the SIP protocol for translation sanity (since it's being NAT'd) and security.
-----Original Message----- From: Josh Patten [mailto:[email protected]] Sent: Monday, January 04, 2010 8:21 PM To: Nathan Nieblas Cc: [email protected] Subject: Re: [sipx-users] multi site deployment http://wiki.sipfoundry.org/display/xecsuserV4r0/Setting+up+BIND+with+loc ation+based+views+for+sipX this may be helpful for you to implement as it creates a much more redundant setup As far as the other issues you are experiencing, make sure your Polycom phones are on a firmware revision no later than 3.1.3c. 3.2 and up has known issues with sipX and should be avoided until sipX version 4.2 is released. Also, are all of your phones showing up in the registration table? What do you mean by "SIP injection"? sipX and sipXbridge generally work best when the SIP messaging isn't messed with by a third party product. Nathan Nieblas wrote: > I come from a Cisco Call Manager background and I am trying to apply the > same centralized concept/design for a sipX deployment... I'm just gonna > throw what I think would be all relevant facts out there and hopefully > one of you guys can point me in the right direction. Thanks in advance. > > Scenario: > 1 Head office, 2 branch offices, 1 Datacenter > All locations are connected by VPN tunnel. > Mixture of Polycom and Cisco phones > PBX located in datacenter, virtualized on ESXi 4.x > > We are randomly having issues with 1-way audio after putting a call on > hold and resuming it or being picked up from call park. The issue > appears to be related with calls that originate from the Auto Attendant. > > Another issue appears to be that Polycom phones cannot dial extensions > on Cisco phones and the call ends up in voicemail but works the other > way around. I have all phones pulling their configuration from the TFTP > server on the sipX box so I'm assuming there shouldn't be any URI/SIP > preferences missing that would keep the phones from talking the same > language. > > This is my digitmap on the Polycom's, our extension ranges are 100-113, > 200-205, 400, 700-705 and 1200-1203: > > [2-9]11|0T|10x|11x|[2-7]xx|120x|*xxxx|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx > |91[2-9]xxxxxxxxx|9[2-9]xxxxxx|[8]xxx > > The PBX assigned a private address and is NAT'd behind a Cisco ASA > that's handling all the SIP inspection, Bandwidth.com is our trunk > provider. I have calls delivering over port 5080 to a local sipXbridge > and I have SIP inspection also being done over that port with the ASA. I > have nothing special configured for inspection. > > What could I be doing wrong here? > > Nathan Nieblas > SACA Technologies, Inc. > > _______________________________________________ > 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/ > _______________________________________________ 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/
