basically, this is how we're setup with this client. The sipx server is in our Philly datacenter. It's not behind NAT and it has a public IP with firewall turned on. All ports closed except the ones I said earlier. Then we have 2 locations. Our client's office, and our office. We have phones in both locations connected to the sipx server.
As for firewall, both offices are behind firewalls and the polycom phones behind NAT. Location A has extensions 5001, 5004. Location B has 5002, 5003. Incoming calls go through without issue for both locations. THey can receive and make calls. We can receive and make calls. However, Location A cannot contact Location B. But we can call from within the same location. 5001 can call 5004 but not 5002. 5002 can call 5003 but not 5001. Although the extensions would ring, except that there is no audio when the call is picked up. But within the same location audio is fine. Also, if 5001 calls 5002 and 5002 does not pickup, the call successfully gets redirected to the VM system with audio working. On Apr 16, 2010, at 11:36 AM, Scott Lawrence wrote: > On Fri, 2010-04-16 at 10:59 -0400, Francis Tinio wrote: >> Hi. >> >> Our issue with calls suddenly disconnecting after 18 secs has been >> fixed. >> >> However, it appears another problem that I initially thought was >> linked to this is not. We cannot seem to call another extension on a >> different location. Nor can we transfer calls between phones on >> different locations. We can however call extensions located in the >> same location and transfer calls in the same location. >> >> Are there specific ports needed for call transferring? I know we have >> enabled ports 21, 8443, 12000, 5060, 5070 5080, 30000-31000 > > There's no way to answer your question without a careful description how > your different locations are configured (dial plans, extension > numbering, local network definitions), what phones you are using > (including version numbers), and how your network is configured (are > there NATs between the locations, private VPNs...). > > After you've described all that... a careful description of what call is > made that does not work, and specifically what failure you see (sip > traces please, not narrative about what noises are heard). > > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org 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/
