Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <[email protected]> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <61369> Message-ID: <[email protected]>
gump103 wrote on Tue, 28 June 2011 08:46 > Hi Hoping someone may be able to point me in the right > direction as this has had me banging my head against the > wall now for a couple of weeks. > > We have a uk ITSP which we use with our current ip phone > solution which works fine but the phone system doesn't > have all the features of sipxecs so I am really keen to > move us over to a sipxecs solution. > > I've got all the internal stuff working, with HA, AD > intergration, mybuddy etc working fine but have a problem > getting the external ITSP to work properly. The trunk > establishes and calls can be made (incoming and outgoing) > but incoming calls disconnect after 32 seconds and > outgoing calls (and incoming if less than 32 seconds) > don't see if the external person hangs up. > > Then the call is not being ACK'ed. > > Initially I was using a pfsense firewall for the test > and was having the 32 second cut off which was caused > (according to the supplier) by nat changing the port but > sipxecs asking for the port to change to 5080, I then > changed to a spare production router which is the same as > our live one and had exacly the same problem except for > the fact that the ports are now right. I have tried > forwarding all external ports without any difference. > > I have also configure a freepbx (asterisk) box to > confirm that the test account was working and it worked > fine first time and I am now being told by the itsp that > it is most likely the sipxecs product and I shouldn't be > looking at it. > > I have tried turning alg on and off and set outgoing > mappings for port 5080 and 5060 for nat but have > completely run out of ideas. > > Any suggestions will be gratefully recieved. I can also > supply any logs that might be of use to help me. > > thanks > James Does the ITSP not allow calls to be sent on port 5080 from them to you? You will still be sending the calls to them on 5060. All sipx cares about is the invite on inbound calls here. Instead of banging your head against a firewall or sipx, ask the ITSP point blank. 1: If I register to you on port 5080, can you send me call on port 5080? 2: if not, can you configure your system to send calls to me on port 5080 manually? NOW -- If the ITSP says they can/are sending on port 5080, and this is still happening, you need to get a siptrace and see why the call is failing (to ack). Can you get the ITSP to point blank admit whether they can or cant send you calls on port 5080? -- ===================== Tony Graziano, Manager Telephone: 434.984.8430 sip: mailto:[email protected][/email][/email] Fax: 434.984.8431 Email: mailto:[email protected][/email][/email] _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
