# SIP REFER support in Asterisk **URL:** \ **Category:** Asterisk SIP\ **Created:** [April 16, 2019, 6:26am UTC](https://community.asterisk.org/t/sip-refer-support-in-asterisk/79362 "2019-04-16T06:26:49Z")\ **Posts on this page:** 1\ **Showing post:** 2
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [April 16, 2019, 8:08am UTC](https://community.asterisk.org/t/sip-refer-support-in-asterisk/79362/2 "2019-04-16T08:08:43Z")
What are SIP trunks. There is no such concept in SIP or Asterisk. Asterisk supports REFER and REFER/Replaces when acting as a UAS, whether the peer is a phone or another PABX. It also supports REFER, but not REFER/Replaces, as a UAC. Again it doesn’t care whether the peer is a phone or another PABX. This is for chan\_sip, but I assume that the same is true for PJSIP. Asterisk doesn’t automatically de-trombone calls. If it receives REFER when acting as a UAS, it will not forward REFER upstream, but will rather just brdige the call internally. You have to use the Transfer application, rather than DIal, to send a REFER. --- _[View the full topic](https://community.asterisk.org/t/sip-refer-support-in-asterisk/79362)._