Sip refer

I already wrote a message but maybe the problem I described was pretty unclear, so here a new Try. The thing is it is really urgent and I am not capable to resolve it for myself. Actually I am not a Asterisk Specialist, but I have to use an Asterisk Server for my thesis…
So here the problem in a way breaked down to core-problem:

SIP-Phone1 registered in Asterisk context: [default]
SIP-Phone2 registered in SIPServerX

SIP-Phone1 <—Call—> SIP-Phone2

SIP-Phone1 --REFER, with–> Asterisk

Now how could I find the “REFER” in the Dialplan!?

That I can do:

exten-> ???,1,do_something(…)

and the “do_something”, could be anything, or should it be “Dial” or “Transfer”?

and what might be the ???

Thank you very much,


If you have canreinvite=yes (or not set) plus a couple of other conditions (such as progressinband=no or unset), transfer will send refer-to. Dial will always use INVITE.

Hello valley,
thank you for your response :smile:
But I fear that this not yet resolve my problem:
Asterisk should not send a REFER, it should HANDLE an incoming REFER.

As you could see in the example below SIP-Phone1 sends an REFER to Asterisk, referring to

I inserted as a peer in sip.conf with context = default

And I wrote in the extensions.conf:

exten => laura,1,Dial(SIP/laura,60)

But when I send the REFER to Asterisk it first answers me with a “REFER accepted”-Message and then sends me NOTIFY, saying:
“SIP/2.0 481 Call leg/transaction does not exist”

Any Idea!?



I’m not a SIP expert, although that’s a little strange for me. My understanding is, Asterisk should simply relay REFER from SIP-Phone1 to SIP-Phone2 in the scenario you originally decribed, not try to handle a SIP message. SIP protocol specifically says that a SIP proxy should not attempt to act on REFER - I think in your scenario, Asterisk’s function is a proxy.

The only situation in which Asterisk really should handle REFER is when Asterisk answers the SIP call and before it hands the call over to another party.

wotanunddasoh are you sure that asterisk don’t send SIP REFER message? i have same problem . my problem is here help me


Asterisk is a back to back user agent. Normally. fpr REFER without replaces, it starts a new dialplan thread in the same was as for an INVITE, but using the original incoming channel.

REFER with replaces is also handled internally by default, but I think you can specify which domains are local, in which case a replace for a call which is not actually local and also not for a local domain will generated an outgoing INVITE/Replaces. There may be some subtle details, and, as very few people use this, it may be buggy.

Note that some SIP phones do INVITE on a second call, followed by REFER/Replaces, even though the user thinks they are doing a blind transfer.