|
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 |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
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 |
|
|
|
|
|
Medium |
|
|
|
|
|
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 |
|
|
|
|
|
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 |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
Bind Address (bindaddr) must be an IP address.
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
Bind Port (bindport) must be between 1024 and 65535
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
Caller ID into Contact Header
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
Medium |
|
|
|
|
|
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 |
|
|
|
|
|
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 |
|
|
|