There is no STUN involved, i set the static ip instead. I have some new information from my provider. They say, the problem is with the interoperability; they are using asterisk and the problem starts when sipx sends the ringback tone. their server doesn't register the call for 10-15 sec and it doesn't open the signal. or something like this. when i set the call to go directly to the operator, they sad it was fine, because there was no ringback tone involved.
is there a way to get this solved? On Mon, Aug 2, 2010 at 10:44 AM, Michael Picher <[email protected]> wrote: > snom 300 should work fine. i don't use them much though. > > make sure the internet calling section is setup properly. find the NAT > settings and set a static outside IP instead of relying on STUN. > > do a packet capture spanning (monitoring) your Internet firewall port and > see what you see for traffic. this is often the most revealing (and another > reason i like pfsense.... can do it right from the gui). > > Mike > > > On Mon, Aug 2, 2010 at 4:29 AM, Irena Dolovčak > <[email protected]>wrote: > >> >> >> >> Mike, thanks for your reply. >> >> I have no SIP helpers enabled.. The first thing I do is to turn them off.. >> >> So, I have contacted my provider, and they say that their server doesn't >> send anything when media should be sent. But i'm really not sure if that's >> their problem.. I think it could be that they don't get all the information >> they need.. >> OK, there are "just" 4 things that could be wrong.. How could I eliminate >> some thing to get closer to the solution of the problem? >> Either the providers sever, my phone, my firewall or my server. >> I don't think that the firewall is blocking something.. because the server >> from my provider doesn't sent anything.. >> Are there some known issues with Snom 300 phones? >> >> >> On Sat, Jul 31, 2010 at 10:23 AM, Michael Picher <[email protected]>wrote: >> >>> this is usually the fault of the firewall... >>> >>> how the firewall handles outbound NAT can cause problems (specially >>> outbound port randomization). also, of course as has been said MANY times >>> on this list, if you have any SIP helpers in the firewall, turn them off... >>> >>> if you are still having trouble, try pfSense firewall. >>> >>> Mike >>> >>> On Fri, Jul 30, 2010 at 3:23 AM, Irena Dolovčak < >>> [email protected]> wrote: >>> >>>> oh yes, i forgot to tell.. the snom (on the inside) cannot hear the >>>> other side talking.. >>>> >>>> On Fri, Jul 30, 2010 at 8:50 AM, Irena Dolovčak < >>>> [email protected]> wrote: >>>> >>>>> Hi, >>>>> >>>>> I have sucessfully established an outbound call, but when i make an >>>>> inbound call, i got just one way audio.. has anybody an idea? >>>>> >>>>> here is the configuration: >>>>> sipx (local Ip is 192.168.1.6); snom phone (192.168.1.10); router >>>>> (10.160.250.62); thi sip gateway is 10.160.4.146 >>>>> >>>>> i changed the media ports to 50000-51000 >>>>> so i'm really not sure why there is a problem with inbound calls.. i >>>>> couldn't find a reason.. >>>>> >>>>> here are the traces of inbound and outbound calls (on the server) >>>>> >>>>> is there something that i didn't see? >>>>> >>>>> Thanks, >>>>> >>>>> -- >>>>> Irena Dolovčak >>>>> >>>> >>>> >>>> >>>> -- >>>> Irena Dolovčak >>>> >>>> _______________________________________________ >>>> sipx-users mailing list >>>> [email protected] >>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>> >>> >>> >>> >>> -- >>> There are 10 kinds of people in this world, those who understand binary >>> and those who don't. >>> >>> [email protected] >>> blog: http://www.sipxecs.info >>> call: sip:[email protected] <sip%[email protected]> >>> >> >> >> >> -- >> Irena Dolovčak >> >> >> >> -- >> Irena Dolovčak >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > There are 10 kinds of people in this world, those who understand binary and > those who don't. > > [email protected] > blog: http://www.sipxecs.info > call: sip:[email protected] <sip%[email protected]> > -- Irena Dolovčak
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