Asterisk cmd Transfer() after Answer()

Gilberto1 · January 22, 2018, 12:16pm

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… the connection should be answered and the user will enter DTMF and the connection will be transferred with the transfer.

It needs to be with the transfer because it is a sip redirect server.

Works:

exten => s,1,Transfer(SIP/1120@xxx.xxx.xxx.xxx)

Not Works:

exten => s,1,Answer()
...
read some dtmf
...
exten => s,n,Transfer(SIP/1120@xxx.xxx.xxx.xxx)

The error is:

[2018-01-22 07:31:56] NOTICE[32021]: chan_sip.c:20716 handle_response_refer: SIP transfer to <sip:1120@xxx.xxx.xxx.xxx> failed, REFER not allowed.
    -- Auto fallthrough, channel 'SIP/1313001-000e0e7c' status is 'UNKNOWN'

jcolp · January 22, 2018, 12:16pm

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.

david551 · January 22, 2018, 12:23pm

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 reaches the peer), but is refusing the second.

If I understand the diagnostic correctly, it is not even rejecting it, but has announced, at the start of the call, that it will not accept that method of transfer.

Gilberto1 · January 22, 2018, 12:45pm

Good morning!

And what would be the mechanism to accept the transfer after the answer?

Thank you.

david551 · January 22, 2018, 12:58pm

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.

Gilberto1 · January 22, 2018, 1:06pm

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 destruction of SIP dialog '10-02A148C1-5A65DF3B000BC76F-04905700' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:46.19.209.14:5060;transport=udp> for address/port to send to
set_destination: set destination to 46.19.209.14:5060
Reliably Transmitting (NAT) to 46.19.209.14:5060:
BYE sip:46.19.209.14:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111:5060;branch=z9hG4bK2d4e8529;rport
Max-Forwards: 70
From: <sip:DIDWW541152354938@...:5060>;tag=as5e673d59
To: 5412300003 <sip:1313000@46.19.209.14>;tag=125C6E3B-5A65DF3B000BC718-04905700
Call-ID: 10-02A148C1-5A65DF3B000BC76F-04905700
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.15.0
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0

jcolp · January 22, 2018, 1:13pm

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

Gilberto1 · January 22, 2018, 1:23pm

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, channel 'SIP/1313000-0005bba2' status is 'UNKNOWN'
Scheduling destruction of SIP dialog '10-5A1227A7-5A65E48A00003301-04804700' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:46.19.209.14:5060;transport=udp> for address/port to send to
set_destination: set destination to 46.19.209.14:5060
Reliably Transmitting (NAT) to 46.19.209.14:5060:
BYE sip:46.19.209.14:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 187.12.182.140:5060;branch=z9hG4bK042668d0;rport
Max-Forwards: 70
From: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as3e7a124c
To: 5412300002 <sip:1313000@46.19.209.14>;tag=2DDFC988-5A65E48A000032CB-04804700
Call-ID: 10-5A1227A7-5A65E48A00003301-04804700
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.15.0
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0


---

<--- SIP read from UDP:46.19.209.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 187.12.182.140:5060;branch=z9hG4bK042668d0;rport=5060
From: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as3e7a124c
To: 5412300002 <sip:1313000@46.19.209.14>;tag=2DDFC988-5A65E48A000032CB-04804700
Call-ID: 10-5A1227A7-5A65E48A00003301-04804700
CSeq: 103 BYE
Server: DIDWW SBC node
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '10-5A1227A7-5A65E48A00003301-04804700' Method: ACK
Really destroying SIP dialog '0f194f3c05d1bb7123f2c2081805b7ca@10.134.0.7:5060' Method: OPTIONS

<--- SIP read from UDP:10.134.0.5:5060 --->
OPTIONS sip:10.134.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.134.0.5:5060;branch=z9hG4bK710ee396;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.134.0.5>;tag=as45aafd81
To: <sip:10.134.0.1>
Contact: <sip:asterisk@10.134.0.5:5060>
Call-ID: 125952b336f5a54c11db0a252176a6b4@10.134.0.5:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 22 Jan 2018 13:09:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in default (domain 10.134.0.1)

<--- Transmitting (NAT) to 10.134.0.5:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.134.0.5:5060;branch=z9hG4bK710ee396;received=10.134.0.5;rport=5060
From: "asterisk" <sip:asterisk@10.134.0.5>;tag=as45aafd81
To: <sip:10.134.0.1>;tag=as222f53e6
Call-ID: 125952b336f5a54c11db0a252176a6b4@10.134.0.5:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '125952b336f5a54c11db0a252176a6b4@10.134.0.5:5060' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:10.134.0.4:5060 --->
OPTIONS sip:10.134.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.134.0.4:5060;branch=z9hG4bK33592334;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.134.0.4>;tag=as4bbe5d70
To: <sip:10.134.0.1>
Contact: <sip:asterisk@10.134.0.4:5060>
Call-ID: 111d53606b9a9f6563aa84836598f92c@10.134.0.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 22 Jan 2018 13:23:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in default (domain 10.134.0.1)

<--- Transmitting (NAT) to 10.134.0.4:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.134.0.4:5060;branch=z9hG4bK33592334;received=10.134.0.4;rport=5060
From: "asterisk" <sip:asterisk@10.134.0.4>;tag=as4bbe5d70
To: <sip:10.134.0.1>;tag=as57050af2
Call-ID: 111d53606b9a9f6563aa84836598f92c@10.134.0.4:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '111d53606b9a9f6563aa84836598f92c@10.134.0.4:5060' in 32000 ms (Method: OPTIONS)
CONF-OI-ASSIST*CLI>
Disconnected from Asterisk server

david551 · January 22, 2018, 1:26pm

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.

Gilberto1 · January 22, 2018, 1:35pm

<--- 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@servidor140.conferenciacorp.com.br:5060>
CSeq: 10 INVITE
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE
Diversion: <sip:+541152354938@sip.didww.com>;reason=unconditional
User-Agent: DIDWW SBC node
Content-Type: application/sdp
Contact: <sip:46.19.209.14:5060;transport=udp>
Content-Length: 298

v=0
o=- 1476805923 2693868391 IN IP4 46.19.209.79
s=-
t=0 0
m=audio 32622 RTP/AVP 0 8 18 4 101
c=IN IP4 46.19.209.79
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=direction:both
<------------->
--- (13 headers 14 lines) ---
Sending to 46.19.209.14:5060 (NAT)
Using INVITE request as basis request - 10-54BD44B6-5A65E7C80003B63A-FCE0A700
Found peer '1313000' for '1313000' from 46.19.209.14:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100 (g729), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 46.19.209.79:32622
Looking for DIDWW541152354938 in from-didww (domain servidor140.conferenciacorp.com.br)
list_route: hop: <sip:46.19.209.14:5060;transport=udp>
RDNIS for this call is +541152354938 (reason unconditional)

<--- Transmitting (NAT) to 46.19.209.14:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 46.19.209.14;branch=z9hG4bKPIGZyaXY;received=46.19.209.14;rport=5060
From: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700
To: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
CSeq: 10 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:DIDWW541152354938@187.12.182.140:5060>
Content-Length: 0


<------------>
    -- Executing [DIDWW541152354938@from-didww:1] Goto("SIP/1313000-0005bbb4", "from-didww10-argentina,s,1") in new stack
    -- Goto (from-didww10-argentina,s,1)
    -- Executing [s@from-didww10-argentina:1] Answer("SIP/1313000-0005bbb4", "") in new stack
Audio is at 14324
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 46.19.209.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 46.19.209.14;branch=z9hG4bKPIGZyaXY;received=46.19.209.14;rport=5060
From: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700
To: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as4fe2605c
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
CSeq: 10 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:DIDWW541152354938@187.12.182.140:5060>
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 361730092 361730092 IN IP4 187.12.182.140
s=Asterisk PBX 1.8.15.0
c=IN IP4 187.12.182.140
t=0 0
m=audio 14324 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:46.19.209.14:5060 --->
ACK sip:DIDWW541152354938@187.12.182.140:5060 SIP/2.0
Via: SIP/2.0/UDP 46.19.209.14;branch=z9hG4bK9C5HjagA;rport
From: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700
To: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as4fe2605c
CSeq: 10 ACK
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
Max-Forwards: 70
Contact: <sip:46.19.209.14:5060;transport=udp>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- Executing [s@from-didww10-argentina:2] Wait("SIP/1313000-0005bbb4", "2") in new stack
    -- Executing [s@from-didww10-argentina:3] Set("SIP/1313000-0005bbb4", "CDR(userfield)=ARGENTINA10-DIDWW") in new stack
    -- Executing [s@from-didww10-argentina:4] Set("SIP/1313000-0005bbb4", "MAX_ERRO_SALA=3") in new stack
    -- Executing [s@from-didww10-argentina:5] Set("SIP/1313000-0005bbb4", "MAX_ERRO_SENHA=3") in new stack
    -- Executing [s@from-didww10-argentina:6] Set("SIP/1313000-0005bbb4", "TIMEOUT(absolute)=21600") in new stack
Channel will hangup at 2018-01-22 17:27:20.773 BRST.
    -- Executing [s@from-didww10-argentina:7] Set("SIP/1313000-0005bbb4", "CHANNEL(language)=es") in new stack
    -- Executing [s@from-didww10-argentina:8] Transfer("SIP/1313000-0005bbb4", "SIP/1148412001@200.196.248.156") in new stack
set_destination: Parsing <sip:46.19.209.14:5060;transport=udp> for address/port to send to
set_destination: set destination to 46.19.209.14:5060
Reliably Transmitting (NAT) to 46.19.209.14:5060:
REFER sip:46.19.209.14:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 187.12.182.140:5060;branch=z9hG4bK539039e3;rport
Max-Forwards: 70
From: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as4fe2605c
To: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700
Contact: <sip:DIDWW541152354938@187.12.182.140:5060>
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
CSeq: 102 REFER
User-Agent: Asterisk PBX 1.8.15.0
Refer-To: <sip:1148412001@200.196.248.156>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Referred-By: <sip:DIDWW541152354938@187.12.182.140:5060>
Content-Length: 0


---
Really destroying SIP dialog '7a65947604b3d2a07c5fd9fc5863a0e1@10.134.0.3:5060' Method: OPTIONS

<--- SIP read from UDP:46.19.209.14:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 187.12.182.140:5060;branch=z9hG4bK539039e3;rport=5060
From: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as4fe2605c
To: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
CSeq: 102 REFER
Server: DIDWW SBC node
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[2018-01-22 11:27:21] NOTICE[30548]: chan_sip.c:20716 handle_response_refer: SIP transfer to <sip:1148412001@200.196.248.156> failed, REFER not allowed.
    -- Auto fallthrough, channel 'SIP/1313000-0005bbb4' status is 'UNKNOWN'
Scheduling destruction of SIP dialog '10-54BD44B6-5A65E7C80003B63A-FCE0A700' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:46.19.209.14:5060;transport=udp> for address/port to send to
set_destination: set destination to 46.19.209.14:5060
Reliably Transmitting (NAT) to 46.19.209.14:5060:
BYE sip:46.19.209.14:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 187.12.182.140:5060;branch=z9hG4bK6032a022;rport
Max-Forwards: 70
From: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as4fe2605c
To: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.15.0
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0


---

<--- SIP read from UDP:46.19.209.14:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 187.12.182.140:5060;branch=z9hG4bK6032a022;rport=5060
From: <sip:DIDWW541152354938@servidor140.conferenciacorp.com.br:5060>;tag=as4fe2605c
To: 5434500165 <sip:1313000@46.19.209.14>;tag=2BDFC5FC-5A65E7C80003B5E3-FCE0A700
Call-ID: 10-54BD44B6-5A65E7C80003B63A-FCE0A700
CSeq: 103 BYE
Server: DIDWW SBC node
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '10-54BD44B6-5A65E7C80003B63A-FCE0A700' Method: ACK

<--- SIP read from UDP:200.196.248.156:5060 --->
OPTIONS sip:187.12.182.140 SIP/2.0
Via: SIP/2.0/UDP 200.196.248.156:5060;branch=z9hG4bK7177ffa0;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@200.196.248.156>;tag=as42baf700
To: <sip:187.12.182.140>
Contact: <sip:asterisk@200.196.248.156:5060>
Call-ID: 10f7ea213027f19724cd40776d1c8b00@200.196.248.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX GIT-13-13.15.0-rc1-779-gd2fb0ffM
Date: Mon, 22 Jan 2018 13:31:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in default (domain 187.12.182.140)

<--- Transmitting (NAT) to 200.196.248.156:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 200.196.248.156:5060;branch=z9hG4bK7177ffa0;received=200.196.248.156;rport=5060
From: "asterisk" <sip:asterisk@200.196.248.156>;tag=as42baf700
To: <sip:187.12.182.140>;tag=as54aedb16
Call-ID: 10f7ea213027f19724cd40776d1c8b00@200.196.248.156:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '10f7ea213027f19724cd40776d1c8b00@200.196.248.156:5060' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '41b3f5446138bb823eee96b81fadb86f@10.134.0.2:5060' Method: OPTIONS
CONF-OI-ASSIST*CLI>
Disconnected from Asterisk server

Gilberto1 · January 22, 2018, 1:36pm

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 200.196.248.156:5060;branch=z9hG4bK5b0b64cc;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@200.196.248.156>;tag=as4cd1b722
To: <sip:187.12.182.140>
Contact: <sip:asterisk@200.196.248.156:5060>
Call-ID: 578d9fcd0d6884b85b351381763d58ad@200.196.248.156:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX GIT-13-13.15.0-rc1-779-gd2fb0ffM
Date: Mon, 22 Jan 2018 13:34:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:187.12.182.140:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 200.196.248.156:5060;branch=z9hG4bK5b0b64cc;received=200.196.248.156;rport=5060
From: "asterisk" <sip:asterisk@200.196.248.156>;tag=as4cd1b722
To: <sip:187.12.182.140>;tag=as0132bf8d
Call-ID: 578d9fcd0d6884b85b351381763d58ad@200.196.248.156:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '578d9fcd0d6884b85b351381763d58ad@200.196.248.156:5060' Method: OPTIONS
VIVO156*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

Gilberto1 · January 22, 2018, 1:37pm

Is there an alternative?

jcolp · January 22, 2018, 1:38pm

The entity at 46.19.209.14:5060 which has a user agent of “DIDWW SBC node” does not support you sending a REFER (to do a blind transfer). You can’t use Transfer after answer with them. The only alternative is to not do it.

david551 · January 22, 2018, 1:38pm

I meant Allow. The peer only supports transfers before answer, you will need to talk to the people that support the peer, to see if there is any option to change this.

jcolp · July 25, 2019, 10:24am

A post was split to a new topic: Transfer after Answer in dialplan

Topic		Replies	Views
SIP REFER issue Asterisk Support	3	253	June 10, 2011
Redirecting/Transferring an answered call Asterisk SIP	7	51	December 4, 2024
Transfer after Answer in dialplan Asterisk Support	9	1498	August 24, 2019
Problem with transfer Asterisk Support	5	894	February 19, 2013
Call transfer with call back to transferer Asterisk Support	8	833	August 30, 2012

Asterisk cmd Transfer() after Answer()

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