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

Please note

• There are SIP clients that do not work well with these settings, like the Cisco ATA 186.

• See Asterisk sip canreinvite for more information.

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