Source Priority Failing checks
If you clear each codec and then add them one at a time, submitting with each addition, they will be added in order which will effect the codec priority.
Medium
See current version of Asterisk for limitations on SRV functionality.
Medium
Comment
%s must be a non-negative integer
Medium
%s must be alphanumeric
Medium
Adaptive
Medium
Add Address
Medium
Add Field
Medium
Add Local Network Field
Medium
Advanced General Settings
Medium
After you enable/disable a transport, asterisk needs to be <strong>restarted</strong>, not just reloaded.
Medium
Allow Anonymous Inbound SIP Calls
Medium
Allow SIP Guests
Medium
Allow Transports Reload
Medium
Allow transports to be reloaded when the PBX is reloaded. Enabling this is not recommended, and may lead to issues.
Medium
Allowing Inbound Anonymous SIP calls means that you will allow any call coming in form an un-known IP source to be directed to the 'from-pstn' side of your dialplan. This is where inbound calls come in. Although FreePBX severely restricts access to the internal dialplan, allowing Anonymous SIP calls does introduced additional security risks. If you allow SIP URI dialing to your PBX or use services like ENUM, you will be required to set this to Yes for Inbound traffic to work. This is NOT an Asterisk sip.conf setting, it is used in the dialplan in conjuction with the Default Context. If that context is changed above to something custom this setting may be rendered useless as well as if 'Allow SIP Guests' is set to no.
Medium
An Error occurred trying fetch network configuration and external IP address
Medium
An unknown port conflict has been detected in PJSIP. Please check and validate your PJSIP Ports to ensure they're not overlapping
Medium
Asterisk NAT setting:<br /> yes = Always ignore info and assume NAT<br /> no = Use NAT mode only according to RFC3581 <br /> never = Never attempt NAT mode or RFC3581 <br /> route = Assume NAT, don't send rport
Medium
Asterisk SIP Settings
Medium
Asterisk is currently using %s for SIP Traffic.
Medium
Asterisk: bindaddr. The IP address to bind to and listen for calls on the Bind Port. If set to 0.0.0.0 Asterisk will listen on all addresses. It is recommended to leave this blank.
Medium
Asterisk: canreinvite. yes: standard reinvites; no: never; nonat: An additional option is to allow media path redirection (reinvite) but only when the peer where the media is being sent is known to not be behind a NAT (as the RTP core can determine it based on the apparent IP address the media arrives from; update: use UPDATE for media path redirection, instead of INVITE. (yes = update + nonat)
Medium
Asterisk: externrefresh. How often to lookup and refresh the External Host FQDN, in seconds.
Medium
Asterisk: g726nonstandard. If the peer negotiates G726-32 audio, use AAL2 packing order instead of RFC3551 packing order (this is required for Sipura and Grandstream ATAs, among others). This is contrary to the RFC3551 specification, the peer _should_ be negotiating AAL2-G726-32 instead.
Medium
Asterisk: t38pt_udptl. Enables T38 passthrough which makes faxes go through Asterisk without being processed.<ul><li>No - No passthrough</li><li>Yes - Enables T.38 with FEC error correction and overrides the other endpoint's provided value to assume we can send 400 byte T.38 FAX packets to it.</li><li>Yes with FEC - Enables T.38 with FEC error correction</li><li>Yes with Redundancy - Enables T.38 with redundancy error correction</li><li>Yes with no error correction - Enables T.38 with no error correction.</li></ul>
Medium
Audio Codecs
Medium
Auto Configure
Medium
Bind Address
Medium
Bind Address (bindaddr) must be an IP address.
Medium
Bind Port
Medium
Bind Port (bindport) must be between 1024 and 65535
Medium
CA Chain File
Medium
CHANSIP Port Moved
Medium
CHANSIP TCP Disabled
Medium
CHANSIP TLS Port Moved
Medium
Call Events
Medium
Caller ID into Contact Header
Medium
Candidates
Medium
Certificate File
Medium
Certificate Manager
Medium
Chan PJSIP Settings
Medium
Chan SIP
Medium
Chan SIP Settings
Medium
Chansip was assigned a port that was already in use for TLS traffic. The Chansip TLS port has been changed to %s
Medium
Chansip was assigned the same port as pjsip for TCP traffic. Chansip has had the tcpenable setting removed, and is no longer listening for TCP connections.
Medium
Chansip was assigned the same port as pjsip for UDP traffic. The Chansip port has been changed to %s
Medium
Check to enable and then choose allowed codecs.
Medium
Codecs
Medium
Control whether subscriptions INUSE get sent ONHOLD when call is placed on hold. Useful when using BLF.
Medium
Control whether subscriptions already INUSE get sent RINGING when another call is sent. Useful when using BLF.
Medium