Usually an audio NAT problem is going to be either firewall or improper settings in NAT/Internet Calling (Admin GUI -> System -> Internet Calling, and Admin GUI -> System -> Servers -> NAT [Left side menu]).
Don't rely on STUN, change to Static outside address... Mike On Sun, Sep 18, 2011 at 10:30 PM, Ewan McLean <[email protected]>wrote: > I tried enabling some NAT stuff in the Cisco config file and got it to dial > out but there was no audio either way and it wasn't really working at all. > Does this indicate a port problem? Tony previously helped me solve a > different issue by disabling the port forwarding on my router so I don't > want to mess with that. Any ideas? Or places in the UK to find polycom > phones at a competitive price? :) > > -- > <http://servistech.co.uk> Ewan McLean > > 02034684428 > servistech.co.uk > > > ------------------------------ > > Ewan McLean <[email protected]> > 15 September 2011 14:30 > > I agree with you I wish I had never picked this up but I got it cheap and > didn't realise how big a pain it is. Looks like polycom phones are running > upwards of £200 for a decent model which I can't throw away unless I could > sell on this one. > > I tried factory resetting the phone and letting sipX do it via TFTP but > although I can dial internally, voicemail etc. it doesn't seem to want to > dial out by dialplan at all. Anyone know where to even start debugging this? > Is it a case of merging the logs and using sipviewer? > > > > ------------------------------ > > Becker, Jesse <[email protected]> > 13 September 2011 13:02 > > I could not agree with Tony more. I spent soooooo much time working with > others in 2008 and 2009 working on the Cisco Plus configurations only to > come to the conclusion that Cisco SIP really isn't standard SIP. There are > so many interop issues with their SIP firmware. It only works right if you > use call manager. You should see all the settings you have to change to get > their CUCM "sip trunk" to work with SipX. > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > ------------------------------ > > Tony Graziano <[email protected]> > 12 September 2011 23:36 > > I consider the 7900's the anti-sip. > > I have a site with some cisco gear that we got to work, but I am very > reluctant to touch it. > > I think other have some 7960's that they have gotten to work, but as remote > phones they are more problematic if i recall. > > i would imagine you have an old dial plan in the phone that is overriding > the standard config. you should try using sipxconfig to configure the phone > for you like any other before tinkering manually (IMO). > > > > > -- > ====================== > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: [email protected] > Fax: 434.465.6833 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > > Blog: > http://blog.myitdepartment.net > > Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > > Ask about our Internet faxservices! > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > ------------------------------ > > Ewan McLean <[email protected]> > 12 September 2011 23:11 > > Thanks Tony, I can't believe that actually worked! I guess my head is still > in the asterisk 'everything hits the server first' way of doing things. I > don't suppose you've configured 7960s before? Inbound calls work fine but > trying to place an outbound clal doesn't. It seems to be trying to direct > the call to my cellphone to a local extension ie > [email protected] instead of going through the established dial > plan. I did have a problem when i was getting this set up with whether or > not to use IPs/DNS and such so I wonder if it might be related? > > ------------------------------ > *undo this:* There's a home broadband router at .1 with various ports > opened through > the firewall/NAT for SIP. > > *make sure that any sip ALG is off on the home broadband router. * > > sipx is already compensating for that, it will handle the nat, and you are > not letting it. > > > > ------------------------------ > > Ewan McLean <[email protected]> > 12 September 2011 22:43 > > Hello > > I'm trying to resolve a one way inbound audio issue. I have a softphone > (Xlite 4) at IP 192.168.1.2 connecting to sipXecs on 192.168.1.100 > > There's a home broadband router at .1 with various ports opened through the > firewall/NAT for SIP. > > The SIP trunk is outside the network at the ITSP. Outbound calls work > great. Inbound calls result in the cell phone being able to talk to the > softphone but not the other way around. > > I'm confident this is some sort of port/NAT/firewall/config issue but I've > tried looking around online and I keep getting directed to solutions with > different circumstances so any help you can give would be much appreciated!! > > > Ewan > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Michael Picher eZuce Director of Technical Services O.978-296-1005 X2015 M.207-956-0262 @mpicher <http://twitter.com/mpicher> www.ezuce.com
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