I need to transfer a call using the asterisk ‘transfer’ function after the ‘answer’ function answers the call. If I transfer the call without answering with the ‘answer’ works normally. If I transfer after the answer the error at the end of the post is displayed in the cli: My context is simple… t…

You would need to provide the SIP trace (sip set debug on) but based on the message it appears as though the REFER from chan_sip is being rejected by the remote side.

The peer does not support transfers after answer! Blind transfers after answer does work, but it uses a completely different mechanism from that used before answer. Your peer accepts the former method (actually it can’t do much else, as the call has been released from Asterisk before the redirect …

Good morning! And what would be the mechanism to accept the transfer after the answer? Thank you.

Asterisk sends a REFER request to the peer. However the diagnostic appears to say that the peer has announced that it doesn’t support REFER requests.

The Debug: <-------------> --- (8 headers 0 lines) --- [2018-01-22 10:50:52] NOTICE[30548]: chan_sip.c:20716 handle_response_refer: SIP transfer to <sip:1148412001@000.000.000.000> failed, REFER not allowed. -- Auto fallthrough, channel 'SIP/1313000-0005bb7d' status is 'UNKNOWN' Scheduling dest…

Your output is incomplete. It did not include the transfer attempt or call setup.

That’s all that appeared after the transfer attempt. Something is missing? <-------------> --- (8 headers 0 lines) --- [2018-01-22 11:13:30] NOTICE[30548]: chan_sip.c:20716 handle_response_refer: SIP transfer to <sip:1148412001@200.196.248.156> failed, REFER not allowed. -- Auto fallthrough, ch…

If it is being refused because the peer doesn’t support REFER, you need to look at the incoming INVITE that started the call. In particular, the Accept header.

<--- SIP read from UDP:46.19.209.14:5060 ---> INVITE sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060 SIP/2.0 Via: SIP/2.0/UDP 46.19.209.14;branch=z9hG4bKPIGZyaXY;rport From: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700 To: <sip:DIDWW541152354938@servidor…

On the server at the other end this appears: <------------> Scheduling destruction of SIP dialog '690f3d643fc4deeb7a926fca37c60d4f@200.148.138.37:5060' in 32000 ms (Method: OPTIONS) Reliably Transmitting (NAT) to 187.12.182.140:5060: OPTIONS sip:187.12.182.140 SIP/2.0 Via: SIP/2.0/UDP 200.196.248.1…

Asterisk cmd Transfer() after Answer()

Asterisk Asterisk SIP

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Asterisk cmd Transfer() after Answer()

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