03-14-2018 07:12 AM - edited 03-18-2019 01:58 PM
Good morning,
I've recently upgraded an SX20 from TC7.3.6 to CE8.3.4 and am trying to migrate from H323 to SIP-only dialing. I have it pointed to our registrar, which is a VCS, and I get no errors on the diagnostics page. However, when I attempt to place a SIP call via URI, it hangs up and eventually disconnects. I have verified that SIP network services are active. Attached is the result from the call log--I'm hoping someone can give me an idea of where to begin troubleshooting.
Thank you.
Solved! Go to Solution.
03-16-2018 08:02 AM - edited 03-16-2018 08:04 AM
To Zac's point - go to your VCS and view the search history of the call in question. My guess is your search rules are not configured to support whatever it is you're dialing. With H.323 if you dial an alias or a number like "jenny" or 8675309 it will dial just that. With SIP the dialing standard is a full URI - so if you don't dial a URI the endpoint will automatically append your domain to the dial string, my guess is that's your problem. For example if your SX20 is registered as "john@acme.com" and you dial either of my examples above you will see them in the VCS as dialing jenny@acme.com or 8675309@acme.com.
You can use regex to modify the dial string - strip domains, change them, etc before rerouting them to wherever the next hop is, CUCM or whatever.
Hope this helps!
03-14-2018 08:06 AM
If the VCS is you primary call control device, you should start by tracing the call on the VCS.
03-16-2018 08:02 AM - edited 03-16-2018 08:04 AM
To Zac's point - go to your VCS and view the search history of the call in question. My guess is your search rules are not configured to support whatever it is you're dialing. With H.323 if you dial an alias or a number like "jenny" or 8675309 it will dial just that. With SIP the dialing standard is a full URI - so if you don't dial a URI the endpoint will automatically append your domain to the dial string, my guess is that's your problem. For example if your SX20 is registered as "john@acme.com" and you dial either of my examples above you will see them in the VCS as dialing jenny@acme.com or 8675309@acme.com.
You can use regex to modify the dial string - strip domains, change them, etc before rerouting them to wherever the next hop is, CUCM or whatever.
Hope this helps!
03-19-2018 09:35 AM
Thanks!
02-16-2023 08:34 PM
I'm facing the same issue after upgrading the other sx20 can perfectly make a call towards sip but the upgraded sx20 call doesn't go through
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide