I have been provided a SIP trunk IP/port(172.X.X.X, port 5060) to turn up a SIP trunk on my Cisco 2821 router. Currently this is set up in a lab environment. Right now I have the system configured for four FXO incoming lines feeding out to a Cisco IP 7XXX phone system. This has been working fine.
Can anybody provide a sample config on the Cisco 2821 router for configuring the router to using a SIP trunk as incoming connection instead of the FXO ports. From what I understand GigabitEthernet0/1 will be the port I use for my incoming SIP trunk. I have not done this before so I do not really know where to start.
I appreciate all the help that I can get.
I just rebuilt my home 2821 (version 15.1) using a SIP Trunk provider with 2 DIDs.
I can dial in and out using the attached config example. The actual configuration is larger, more DNs and ePhones, but this should be enough to show the process. I also attached a document I wrote some 7 years ago to explain how to set this up.
The ethernet port you use will depend on your network configuration. I have a SonicWALL FW and a port dedicated to my VoIP router, with the proper access and NAT rules to route SIP and RTSP to this port.
Hope it helps!
Thank you for getting back with me yesterday evening.
I messed around with this some today but am still having issues dialing in/out. I know have my SIP itself active per my ISP.
If I attach the config from my Cisco 2821 do you think you could glance over it and possibly see what I am missing. As I mentioned initially. I had this set to use 4 FXO incoming/outbound analog lines and those were working correctly. I think I am just mixing configs for using a SIP trunk with config for using FXO lines. My SIP trunk connects directly back to my ISP through GigabitEthernet0/1 so I am not routing back to a router or anything. It is staying on a local direct connection between myself and ISP. The IP address that my ISP gave me for my SIP trunk was 172.17.63.2 port 5060. I have a PBX assigned to me as 7403717001 with DiDs ranging from 7403717009-7013. Thanks again for any assistance you can provide.
There are no bind statements set on your dial peers to/from your ITSP. Have a look at this document for how to configure this, https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html
It’s not all that much difference between a TDM voice gateway and a SBC (Cube) voice gateway. It’s in the end boils down to having a pair of dial peers of the type VOIP that does the routing of calls instead of a pair that comprises of one POTS and one VOIP. At this link you’d find plenty of documentation for how to configure your router as an SBC. https://www.cisco.com/c/en/us/support/unified-communications/unified-border-element/products-installation-and-configuration-guides-list.html