I am having a hard time trying to remove asterisk from the middle by doing either an attended or blind transfer, essentially a SIP Refer. I can get a 302 to work but its before the channel enters a stasis app and its done via the dialplan. Once the channel enters the stasis app I have tried to use both /continue and /redirect. With /continue its just hitting a different extension on the dial plan which works with Dial app but keeps asterisk in the middle. I have tried Transfer and BlindTransfer apps but it fails, doesnt even send a sip request unless its not logged.
The channel is not bridged, just a normal channel that enters a stasis app where we play prompts and receive dtmf. Later in the call flow we want to do an attended or blind transfer to remove asterisk from the middle.
Asterisk 22
Pjsip
; extensions_ari.conf
exten = 3500,1,Answer()
same = n,Set(JITTERBUFFER(adaptive)=default)
same = n,Stasis(myStasisApp)
same = n,Hangup()
exten = 3501,1,BlindTransfer(PJSIP/sip:user@ipAddress:port)
same = n,Hangup()
exten = 3501,1,Transfer(PJSIP/transfer-test)
same = n,Hangup()
; pjsip.conf
[transfer-test]
type=endpoint
context=myCustomContext
disallow=all
allow=ulaw
aors=transfer-test-aor
allow_transfer=yes
; tested with or without direct media
[transfer-test-aor]
type=aor
contact=sip:user@ipAddress:port
Extension 3501 in extensions_ari.conf is just showing a couple of ways I tested with no luck.
I can post logs for any desired scenario. What is the proper way to do a SIP refer (some transfer/redirect to remove asterisk from the middle) once a channel enters a stasis app?
jcolp
March 4, 2026, 1:43pm
2
There is no ability to trigger an attended transfer. For a SIP REFER it would be using Transfer() or /redirect as you’ve stated. I would suggest providing actual console output of an attempt including SIP trace (pjsip set logger on) and the ARI interaction.
– Ex. calling dialplan extension directly to show it works as a 302 and at least routable.
[2026-03-04 09:10:36] VERBOSE[967] res_pjsip_logger.c: <— Transmitting SIP response (436 bytes) to UDP:myInternalSoftphoneIp:5060 —>sip:3369586321@myInternalSoftphoneIp ;tag=4767sip:3501@myInternalSoftphoneIp ;tag=59239d4c-c334-4dbd-8255-1803463ace42sip:user@ipAddress:5060
[2026-03-04 09:10:36] VERBOSE[972] res_pjsip_logger.c: <— Received SIP request (378 bytes) from UDP:myInternalSoftphoneIp:5060 —>sip:3501@asteriskInternalIp ;tag=59239d4c-c334-4dbd-8255-1803463ace42sip:3369586321@asteriskInternalIp ;tag=4767
– Ari /continue to dialplan extension 3501 which immediately calls the Transfer app
[2026-03-04 09:21:22] DEBUG[323705] res_http_websocket.c: Writing websocket string of length 827
exten = 3501,1,Transfer(PJSIP/transfer-test)
jcolp
March 4, 2026, 2:41pm
6
I’d also do a test by calling Answer(), Wait(1), Transfer() in the dialplan and eliminate ARI as a factor.
[2026-03-04 09:46:57] DEBUG[967] res_pjsip_session.c: The current inv state is CONFIRMED
The log above was from calling the extension directly. Fails when adding the Answer and it has to do an actual Sip refer instead of 302. Invite was just fine, omitted in the logs.
exten = 3501,1,Answer()
I added the wrong module and the transfer app was silently failing.
res_pjsip_xfer.so vs res_pjsip_refer.so
Still need to do more testing but the extension is now working directly after an answer.
The destination needs to make sense to the remote system. I don’t think you have the option of using a PJSIP endpoint name. The documentation is bad, as it doesn’t adequately define “destination”, but you need a full SIP URI that is meaningful to the upstream system.
There is a request from Asterisk to Twilio and the response from Twilio to Asterisk. It is the latter that has the 5060. Addition: The REFER is being sent the URI specified in the Contact header provided by Twilio, which is correct, from the...
I have confirmed in asterisk 22 at least that a PJSIP endpoint name does work as long as its using a valid sip uri. Tested via ARI /redirect and also in the dial plan transfer app. I agree the documentation is lacking, hopefully this post helps someone in the future.