I come from a Cisco Call Manager background and I am trying to apply the same centralized concept/design for a sipX deployment... I'm just gonna throw what I think would be all relevant facts out there and hopefully one of you guys can point me in the right direction. Thanks in advance.
Scenario: 1 Head office, 2 branch offices, 1 Datacenter All locations are connected by VPN tunnel. Mixture of Polycom and Cisco phones PBX located in datacenter, virtualized on ESXi 4.x We are randomly having issues with 1-way audio after putting a call on hold and resuming it or being picked up from call park. The issue appears to be related with calls that originate from the Auto Attendant. Another issue appears to be that Polycom phones cannot dial extensions on Cisco phones and the call ends up in voicemail but works the other way around. I have all phones pulling their configuration from the TFTP server on the sipX box so I'm assuming there shouldn't be any URI/SIP preferences missing that would keep the phones from talking the same language. This is my digitmap on the Polycom's, our extension ranges are 100-113, 200-205, 400, 700-705 and 1200-1203: [2-9]11|0T|10x|11x|[2-7]xx|120x|*xxxx|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx |91[2-9]xxxxxxxxx|9[2-9]xxxxxx|[8]xxx The PBX assigned a private address and is NAT'd behind a Cisco ASA that's handling all the SIP inspection, Bandwidth.com is our trunk provider. I have calls delivering over port 5080 to a local sipXbridge and I have SIP inspection also being done over that port with the ASA. I have nothing special configured for inspection. What could I be doing wrong here? Nathan Nieblas SACA Technologies, Inc. _______________________________________________ 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/
