Asterisk Letting SIP clients connect directly

Asterisk Letting SIP clients connect directly

Asterisk by default connects all media streams through asterisk to be able to connect various protocols and media to each other. If you have two SIP phones, the media path can be connected directly between the phones without going through Asterisk. Asterisk in this case only handles signalling. It requires that both extensions are using SIP and support the same codecs.

Disable transfer

If you want to transfer calls by pressing the # key during a call, Asterisk will stay in the media stream to be able to listen for # signals. Remove the "tT" from the dial() command to disable this.

Configuration

This is done in ​​sip.conf​​​ by using   canreinvite=yes in the configuration of the SIP extension. This is the default behaviour.

Example

[morgan]  secret=thesweet43  type=friend  host=dynamic  context=sipexts  mailbox=1050  callerid="morgan@yourdomain.com" <1050>  dmtfmode=rfc2833  canreinvite=yes

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