Hi! i have the same problem, but in my case the provider is my Asterisk, and the other two nodes are other Asterisks mine too. If the provider Dial to node1, and node1 does a Transfer to node2, it works. But if node1 made an Answer before transfer, dont. All my ALLOWs has REFER. The no-Answer working:
2019/07/25 13:32:17.675746 170.251.79.80:5060 -> 170.251.79.62:5060
INVITE sip:333@170.251.79.62;user=phone SIP/2.0
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK60f36712
Max-Forwards: 70
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c3f58b1
To: <sip:333@170.251.79.62;user=phone>
Contact: <sip:trunk@170.251.79.80:5060>
Call-ID: 1282301e32d88d88364b82be0ae5a7f3@170.251.79.80:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.27.0
Date: Thu, 25 Jul 2019 09:32:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1681457391 1681457391 IN IP4 170.251.79.80
s=Asterisk PBX 13.27.0
c=IN IP4 170.251.79.80
t=0 0
m=audio 13302 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
2019/07/25 13:32:17.679261 170.251.79.62:5060 -> 170.251.79.80:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK60f36712;received=170.251.79.80
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c3f58b1
To: <sip:333@170.251.79.62;user=phone>
Call-ID: 1282301e32d88d88364b82be0ae5a7f3@170.251.79.80:5060
CSeq: 102 INVITE
Server: Asterisk PBX 16.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:333@170.251.79.62:5060>
Content-Length: 0
2019/07/25 13:32:17.681418 170.251.79.62:5060 -> 170.251.79.80:5060
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK60f36712;received=170.251.79.80
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c3f58b1
To: <sip:333@170.251.79.62;user=phone>;tag=as7a9834e8
Call-ID: 1282301e32d88d88364b82be0ae5a7f3@170.251.79.80:5060
CSeq: 102 INVITE
Server: Asterisk PBX 16.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: Transfer <sip:333@170.251.79.71>
Content-Length: 0
2019/07/25 13:32:17.681688 170.251.79.80:5060 -> 170.251.79.62:5060
ACK sip:333@170.251.79.62;user=phone SIP/2.0
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK60f36712
Max-Forwards: 70
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c3f58b1
To: <sip:333@170.251.79.62;user=phone>;tag=as7a9834e8
Contact: <sip:trunk@170.251.79.80:5060>
Call-ID: 1282301e32d88d88364b82be0ae5a7f3@170.251.79.80:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.27.0
Content-Length: 0
And now the Answer before transfer case:
2019/07/25 13:33:32.192259 170.251.79.80:5060 -> 170.251.79.62:5060
INVITE sip:333@170.251.79.62;user=phone SIP/2.0
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK20610a07
Max-Forwards: 70
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
To: <sip:333@170.251.79.62;user=phone>
Contact: <sip:trunk@170.251.79.80:5060>
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.27.0
Date: Thu, 25 Jul 2019 09:33:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 278
v=0
o=root 1528082092 1528082092 IN IP4 170.251.79.80
s=Asterisk PBX 13.27.0
c=IN IP4 170.251.79.80
t=0 0
m=audio 19562 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
2019/07/25 13:33:32.198060 170.251.79.62:5060 -> 170.251.79.80:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK20610a07;received=170.251.79.80
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
To: <sip:333@170.251.79.62;user=phone>
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 102 INVITE
Server: Asterisk PBX 16.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:333@170.251.79.62:5060>
Content-Length: 0
2019/07/25 13:33:32.200100 170.251.79.62:5060 -> 170.251.79.80:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK20610a07;received=170.251.79.80
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
To: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 102 INVITE
Server: Asterisk PBX 16.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:333@170.251.79.62:5060>
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 1846220969 1846220969 IN IP4 170.251.79.62
s=Asterisk PBX 16.4.0
c=IN IP4 170.251.79.62
t=0 0
m=audio 12202 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
2019/07/25 13:33:32.200586 170.251.79.80:5060 -> 170.251.79.62:5060
ACK sip:333@170.251.79.62:5060 SIP/2.0
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK3bb26487
Max-Forwards: 70
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
To: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
Contact: <sip:trunk@170.251.79.80:5060>
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.27.0
Content-Length: 0
2019/07/25 13:33:32.216796 170.251.79.62:5060 -> 170.251.79.80:5060
REFER sip:trunk@170.251.79.80:5060 SIP/2.0
Via: SIP/2.0/UDP 170.251.79.62:5060;branch=z9hG4bK48c0c1ce
Max-Forwards: 70
From: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
To: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
Contact: <sip:333@170.251.79.62:5060>
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 102 REFER
User-Agent: Asterisk PBX 16.4.0
Date: Thu, 25 Jul 2019 11:33:32 GMT
Refer-To: <sip:333@170.251.79.71>
Referred-By: <sip:333@170.251.79.62:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
2019/07/25 13:33:32.217272 170.251.79.80:5060 -> 170.251.79.62:5060
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 170.251.79.62:5060;branch=z9hG4bK48c0c1ce;received=170.251.79.62
From: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
To: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 102 REFER
Server: Asterisk PBX 13.27.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:trunk@170.251.79.80:5060>
Content-Length: 0
2019/07/25 13:33:32.217569 170.251.79.80:5060 -> 170.251.79.62:5060
NOTIFY sip:333@170.251.79.62:5060 SIP/2.0
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK280b379b
Max-Forwards: 70
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
To: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
Contact: <sip:trunk@170.251.79.80:5060>
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 13.27.0
Event: refer;id=102
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 16
SIP/2.0 200 OK
2019/07/25 13:33:32.217666 170.251.79.62:5060 -> 170.251.79.80:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 170.251.79.80:5060;branch=z9hG4bK280b379b;received=170.251.79.80
From: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
To: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 103 NOTIFY
Server: Asterisk PBX 16.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:333@170.251.79.62:5060>
Content-Length: 0
2019/07/25 13:33:32.218418 170.251.79.62:5060 -> 170.251.79.80:5060
BYE sip:trunk@170.251.79.80:5060 SIP/2.0
Via: SIP/2.0/UDP 170.251.79.62:5060;branch=z9hG4bK0eb63fad
Max-Forwards: 70
From: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
To: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 16.4.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
2019/07/25 13:33:32.218629 170.251.79.80:5060 -> 170.251.79.62:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 170.251.79.62:5060;branch=z9hG4bK0eb63fad;received=170.251.79.62
From: <sip:333@170.251.79.62;user=phone>;tag=as751870eb
To: "trunk" <sip:trunk@170.251.79.80>;tag=as6c034cb0
Call-ID: 7e0ae4cc2f8d7c7878e66223756e4f0a@170.251.79.80:5060
CSeq: 103 BYE
Server: Asterisk PBX 13.27.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0
How can i fix it?