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You likely made the classic mistake of adding your IP address to the 'domains' ACL, causing all calls to get routed to the public context. Your log isn't complete, so I can't say for sure.

What do you mean it doesn't work? Does the sip message show that privacy is set? It is up to the upstream carrier to handle the private call. If they don't handle it properly the caller ID will still show up.

Yes, this is the only way how to reset the stats. mod_callcenter doesn't give any other way to do so.

Install the sqlite3 client and add the following line to your crontab. 0 1 * * * sqlite3 /var/lib/freeswitch/db/callcenter.db "update agents set no_answer_count = 0, calls_answered = 0, last_status_change = strftime('\%s','now');" Note: The %s is escaped for cron. You'll need to remove the...

Assuming you're using the English prompts, check if the module 'mod_say_en' is loaded. fs_cli -x "module_exists mod_say_en" fs_cli -x "load mod_say_en" If you aren't comfortable using the cli, you can use the module manager in Fusion GUI.

You will need to reset them by directly updating the call center database. You can have a cron job to do it daily.

Here's something I quickly wrote (without testing) to give you an idea. <extension name="DID-Time-Condition" continue="true" break="on-false"> <condition regex="any"> <regex field="destination_number" expression="234557890"/> <regex field="destination_number"...

@whut You can write a time condition dialpan that isn't linked to the time condition page.

I think you sent the same logfile as before

You have two outbound routes that match your dialed regex, BulkVS and NANP.1d10. You've set the t38 variables on the NANP.1d10 route, but the call gets routed to the BulkVS route.

The dialplan logs should show why it doesn't get set. There's also a variable 'refuse_t38', but I doubt you set it somewhere.

The log shows freeswitch refusing T38. This means fax_enable_t38 isn't set on the channel.

That's exactly my point. When freeswitch crashes, the socket disconnects, so this isn't an active calls page issue but a freeswitch issue.

You're asking for an easy way; you can 'easily' achieve this with a dialplan if you know how to write XML dialplan

Does it crash only the socket, or does it crash freeswitch? When you click 'record' on the Yealink phone, it sends a SIP request to freeswitch to start recording. If the socket crashes possibly the whole freeswitch crashes.

You should check the freeswitch logs to see why the user_record dialplan doesn't start the recording.

The only way to or conditions in xml dialplan is by using the regex=any condition style. <condition regex="any"> <regex field="destination_number" expression="^01324334832$"/> <regex field="destination_number" expression="^01742743501$"/> <action application="transfer" data="3000...

Freeswitch has this feature it is called "call camping" Here's the link https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Modules/mod-dptools/bridge/Call-Camping_16354681/#docusaurus_skipToContent_fallback

No need for dialplan the simplest is to use extension settings.

You can set proxy_media=true in your dialplan when sending faxes so freeswitch wouldn't care about the media type. Also, freeswitch has clear logs on T38 negotiation and media re-negotiation. If you can post them here I can try to help you.