4.1.0-015463 2009-05-09T14:04:23 ecs-centos5 I have siptrunking and remote NAT traversal configured, and am trying to test features and make sure my configuration is set properly. I do not have any remote workers setup or configured yet, as I am trying to work through this in more than one step.
Questions about sipXbridge and firewall configurations. Assuming there is a static public IP address and your deployment requires: 1. Siptrunking (bandwidth.com) 2. Remote Workers 3. CMC Provisioning (Bria Professional). Bandwidth.com should be configured to send incoming calls on port 5067. Port forwarding for ports 5060, 5067 and 5080 should be pointed to the private IP address of the sipx server through the firewall. Ports 30000-31000 (udp) should be allowed out and inbound should also be pointed to the private IP address of the sipx server through the firewall. port 12000 (tcp) should also be port fowarded to the sipx ip address (cmc provisioning). Port FOrward list: 5060 5067 5080 30000-31000 System>Servers>NAT Specify the public ip address, public port 5060, tp 30000-31000 sipxbridge sbc external port is 5067, public port (blank, is this correct?). Why does siptrunk require username/password in order to run if bandwidth.com does not require this? Is there something I don;t know about my account or is there correction to be made with the template? I'd like to contribute a step by step for this type of scenario to the wiki, but would really like to get everything working and well documented. Currently I am having some issues getting DTMF to work. I am also unable to see that I can call in to leave voicemail for the test account that actually includes the audio. Calls sent directly to the user (DID) to leave a voicemail end up with a "blank" message. The audio is not being recorded or sent properly. If I call in and have the calls hit "operator", I can transfer to the user account, if I press # to leave a message it does not recognize DTMF at all after transferring the call from AA. So this makes me assume that after a transfer from the AA something is incorrect in the way DTMF is being handled after the first transfer. No voicemail is ever generated by the system either. Changing Media Relay aggressiveness has not shown any difference in this. I can hear system prompts just fine. I see this message in sipxbridge log at debug level. "Error in processing request Bridge with the given ID was not found." I do not see an indication in sipxrelay at debig that indicates it receives DTMF after the first transfer. I saw there was an issue with remote worker audio that Ranga posted a patch to, but in my case I'm not using a remote worker. Does anyone see anything inherently wrong with my firewall configuration or sipxbridge/sbc that might cause this? _______________________________________________ sipx-dev mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-dev Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-dev
