Rhon, Are the handsets behind a NAT, ? i've had trouble with these models in that configuration (with the sipx box on a public IP), whereas the 7940/7960's traverse NAT's fine...
so the 7970's can call other handsets, but they cant call the 7970, that's new to me... are the 7970's loading to TFTP and using the DNS settings as per the sipx TFTP/DNS? what i do for my cisco's is use a cisco router as the DHCP and specify the TFTP option, as the handsets are at various sites and talk over a routed adsl network back to the sipx server, i don't have any NAT issues. Regards, Ben On Sun, Mar 28, 2010 at 1:01 PM, Rhon <[email protected]> wrote: > Hi Ben, > > I'm using sip *8.5(3) *firmware. Using Cisco 7970G phone we can establish > an outgoing calls to the internal phones but not able to receive incoming > calls. This is not a problem with our Polycom 650 phones. > > Any thoughts how we can resolve this problem? > > Thanks for your time! > > Regards > > > > On Sat, Mar 27, 2010 at 2:40 PM, Ben Wannan <[email protected]> wrote: > >> I'm using 8.3.5 on mine, fixes the MWI bug, however has a few issues with >> voicemail (still to debug) >> >> is the 7970G talking to sipx and loading a config ? I'm not sure exactly >> where you are stuck, etc. >> >> Regards, >> >> Ben >> >> >> On Sat, Mar 27, 2010 at 3:16 PM, Rhon <[email protected]> wrote: >> >>> Hi, >>> >>> What firmware version did you use so I can start looking for one? >>> >>> Thanks >>> >>> >>> On Sat, Mar 27, 2010 at 2:30 AM, gabriel <[email protected]> wrote: >>> >>>> just add your sipx server in the filed "processNodeName" on "Node Name" >>>> and then it will work. >>>> >>>> it took a while for me to find a fw that doesn't have issues so good >>>> luck with that ;) >>>> >>>> -gabriel >>>> >>>> >>>> >>>> On Fri, 26 Mar 2010, Rhon wrote: >>>> >>>> Hi, >>>>> >>>>> Thanks for the link. I've already seen it and will try the soonest. In >>>>> the meantime I need a working SEP.xml file for Cisco 7970G phone. SipXecs >>>>> is >>>>> unable to >>>>> load the phone by adding managed phones. >>>>> I'm unsure why at this moment. >>>>> >>>>> Anyone here got this working with SipXecs-4.0.4. >>>>> >>>>> Thanks >>>>> >>>>> On Thu, Mar 25, 2010 at 7:35 PM, Scott Lawrence <[email protected]> >>>>> wrote: >>>>> On Thu, 2010-03-25 at 12:09 +0800, Rhon wrote: >>>>> > Also, how can we trace logs in sipxecs when establishing a call? >>>>> We >>>>> > cannot see what's happening in the backend. All outgoing calls >>>>> didn't >>>>> > get through and we don't have any idea where to look at. >>>>> See: >>>>> >>>>> >>>>> http://wiki.sipfoundry.org/display/xecsuserV4r0/Display+SIP+message+flow+using+Sipviewer >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>> >>> _______________________________________________ >>> 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/ >>> >> >> >
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