This would appear to confirm that on line 1328 it did not not know what your extensionnumber is, presumably that would be also your AMPUSER which would explain why *97 believes that there is no voicemailbox to short circuit.
I think this is anomalous behavior we just don’t see on the more mainline channels like chan_sip and chan_pjsip so harder for us to disentangle WTF.
There’s no proper user set in the FROM header, which is where it will default to grabbing the user details from. I’m not sure how you’ve configured either the phones or Bria but they are not configured right at all.
If this is your first freePBX experience, using Cisco phones with the SCCP module is suicide…
Use any SIP-compatible phone first and learn how to use freePBX with the built-in modules.
There are many sip-phones you can configure using their web-gui. In freePBX you just have to add a pjsip extension.
Or…if you buy a Sangoma phone, EPM is free.
There are many companies that provide free provisioning for their IP phones. To name a few besides Sangoma, there is also ClearlyIP (has a FreePBX provisioning module) and Grandstream (has web-based GDMS provisioning system.) There is also an open-source provisioning module for FreePBX but none of these modules will support SCCP phones unless you convert them to SIP firmware.
Regarding SCCP as a protocol, it has been relegated to outdated phones as Cisco has largely embraced SIP for their newer phones. SCCP does not typically support multiple phones on one extension unless you define each extension as a secondary “line” button on a primary phone. Each phone must have its own primary extension and other “lines” can be defined on a button. The older Cisco phones are cheap because they are nearly impossible to work with on non-Cisco systems. If you try to mix a SIP extension (such as a softphone) with the SCCP extension, you will have many problems.
Since you are using a third-party SCCP protocol for your phones, I’d suggest you go to the forum for the SCCP modules you are using and see if they can offer more assistance.
Yes they can but they will not have the same functionality provided in a Cisco environment. Unless you are running the newer phones like the 8800 series with 3PCC functionality for SIP, you won’t get the features you get with current SIP phones that are built for SIP.
Feel free to beat your head against the wall. Personally, I’ve been there, done that, and won’t do it again. I have long since thrown my Cisco 7900 series phones in the recycle bin (along with my Nortel and Avaya phones.) Proprietary protocol phones will never work well with SIP and Asterisk.
In your config file, try changing the dateTemplate to M/D/YA
This website is extremely handy for customizing your configuration files but also addresses how to set things up if you load SIP firmware on the phones: https://usecallmanager.nz/sepmac-cnf-xml.html