Hello, we are having an issue with attended transfers where once the call is transferred, the destination user will hear music on hold still being played. The other party can hear the destination however, the destination only will hear the music on hold.
Blind transfers work without issues.
Asterisk 18.9.0
some SIP / Asterisk logs:
<--- Received SIP request (990 bytes) from UDP:4.4.255.171:5060 --->
REFER sip:4.4.255.244:5060 SIP/2.0
Via: SIP/2.0/UDP 4.4.255.171;branch=z9hG4bK1e47.d28c166220610cecb995e49d6776bd26.0
Via: SIP/2.0/UDP x.x.80.157:5060;received=x.x.80.157;branch=z9hG4bK426355810;rport=5060
From: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
To: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
CSeq: 1493 REFER
Contact: <sip:MBt8Jz59T9HL@x.x.80.157:5060>
X-Grandstream-PBX: true
Max-Forwards: 69
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.57
Refer-To: <sip:100@97c5b363.sip.sipdomain.net?Replaces=2039913333-5060-203%40BJC.BGI.B.JB%3Bto-tag%3Dd9f53c4c-210d-4dfb-a905-ed495d4e1c46%3Bfrom-tag%3D1161355585>
Referred-By: <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
X-AccountDomain: 97c5b363.sip.sipdomain.net
<--- Transmitting SIP response (723 bytes) to UDP:4.4.255.171:5060 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 4.4.255.171;rport=5060;received=4.4.255.171;branch=z9hG4bK1e47.d28c166220610cecb995e49d6776bd26.0
Via: SIP/2.0/UDP x.x.80.157:5060;rport=5060;received=x.x.80.157;branch=z9hG4bK426355810
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
From: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
To: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
CSeq: 1493 REFER
Expires: 600
Contact: <sip:4.4.255.244:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 18.9.0
Content-Length: 0
<--- Transmitting SIP request (705 bytes) to UDP:4.4.255.171:5060 --->
NOTIFY sip:MBt8Jz59T9HL@x.x.80.157:5060 SIP/2.0
Via: SIP/2.0/UDP 4.4.255.244:5060;rport;branch=z9hG4bKPj0c8dc465-cc2a-400a-a2c7-a80938f81d35
From: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
To: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
Contact: <sip:4.4.255.244:5060>
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
CSeq: 24737 NOTIFY
Route: <sip:4.4.255.171;lr;did=07b.08f2>
Event: refer
Subscription-State: active;expires=600
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 18.9.0
Content-Type: message/sipfrag;version=2.0
Content-Length: 20
SIP/2.0 100 Trying
<--- Transmitting SIP request (711 bytes) to UDP:4.4.255.171:5060 --->
NOTIFY sip:MBt8Jz59T9HL@x.x.80.157:5060 SIP/2.0
Via: SIP/2.0/UDP 4.4.255.244:5060;rport;branch=z9hG4bKPjd39f839c-0ee1-4aa5-a114-6ddcf774236d
From: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
To: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
Contact: <sip:4.4.255.244:5060>
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
CSeq: 24738 NOTIFY
Route: <sip:4.4.255.171;lr;did=07b.08f2>
Event: refer
Subscription-State: terminated;reason=noresource
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 18.9.0
Content-Type: message/sipfrag;version=2.0
Content-Length: 16
SIP/2.0 200 OK
-- Stopped music on hold on PJSIP/sbc01-outbound-00010129
-- Channel PJSIP/sbc01-outbound-00010129 left 'simple_bridge' basic-bridge <ea1bcce4-939c-4160-a6e2-7109b4c60d81>
-- Channel PJSIP/km01-00010128 left 'simple_bridge' basic-bridge <ea1bcce4-939c-4160-a6e2-7109b4c60d81>
<PJSIP/km01-00010128>AGI Tx >> 200 result=-1
<--- Transmitting SIP request (503 bytes) to UDP:4.4.255.171:5060 --->
BYE sip:MBt8Jz59T9HL@x.x.80.157:5060 SIP/2.0
Via: SIP/2.0/UDP 4.4.255.244:5060;rport;branch=z9hG4bKPjb578b0e4-fa50-49b2-b612-2121d91b69a6
From: <sip:100@97c5b363.sip.sipdomain.net>;tag=d9f53c4c-210d-4dfb-a905-ed495d4e1c46
To: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1161355585
Call-ID: 2039913333-5060-203@BJC.BGI.B.JB
CSeq: 4423 BYE
Route: <sip:4.4.255.171;lr;did=418.3b6>
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.9.0
Content-Length: 0
-- <PJSIP/km01-00010128>AGI Script /etc/asterisk/callflow_handler.py completed, returning 0
-- Auto fallthrough, channel 'PJSIP/km01-00010128' status is 'ANSWER'
<--- Received SIP response (657 bytes) from UDP:4.4.255.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.4.255.244:5060;received=4.4.255.244;rport=5060;branch=z9hG4bKPj0c8dc465-cc2a-400a-a2c7-a80938f81d35
Record-Route: <sip:4.4.255.171;lr;did=07b.08f2>
From: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
To: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
CSeq: 24737 NOTIFY
Contact: <sip:MBt8Jz59T9HL@x.x.80.157:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<--- Received SIP response (657 bytes) from UDP:4.4.255.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.4.255.244:5060;received=4.4.255.244;rport=5060;branch=z9hG4bKPjd39f839c-0ee1-4aa5-a114-6ddcf774236d
Record-Route: <sip:4.4.255.171;lr;did=07b.08f2>
From: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
To: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
CSeq: 24738 NOTIFY
Contact: <sip:MBt8Jz59T9HL@x.x.80.157:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<--- Received SIP response (595 bytes) from UDP:4.4.255.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.4.255.244:5060;received=4.4.255.244;rport=5060;branch=z9hG4bKPjb578b0e4-fa50-49b2-b612-2121d91b69a6
From: <sip:100@97c5b363.sip.sipdomain.net>;tag=d9f53c4c-210d-4dfb-a905-ed495d4e1c46
To: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1161355585
Call-ID: 2039913333-5060-203@BJC.BGI.B.JB
CSeq: 4423 BYE
Contact: <sip:MBt8Jz59T9HL@x.x.80.157:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<--- Received SIP request (721 bytes) from UDP:4.4.255.171:5060 --->
BYE sip:4.4.255.244:5060 SIP/2.0
Via: SIP/2.0/UDP 4.4.255.171;branch=z9hG4bKed47.a700ad074962f3ff0e5d8e78b340a342.0
Via: SIP/2.0/UDP x.x.80.157:5060;received=x.x.80.157;branch=z9hG4bK1356464725;rport=5060
From: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
To: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
CSeq: 1494 BYE
Contact: <sip:MBt8Jz59T9HL@x.x.80.157:5060>
X-Grandstream-PBX: true
Max-Forwards: 69
Supported: replaces, path, timer
User-Agent: Grandstream GXP2160 1.0.11.57
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<--- Transmitting SIP response (506 bytes) to UDP:4.4.255.171:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.4.255.171;rport=5060;received=4.4.255.171;branch=z9hG4bKed47.a700ad074962f3ff0e5d8e78b340a342.0
Via: SIP/2.0/UDP x.x.80.157:5060;rport=5060;received=x.x.80.157;branch=z9hG4bK1356464725
Call-ID: 87132468-5060-202@BJC.BGI.B.JB
From: "101" <sip:MBt8Jz59T9HL@97c5b363.sip.sipdomain.net>;tag=1043365788
To: <sip:3052990233@97c5b363.sip.sipdomain.net>;tag=2c3493a3-8a29-4d0b-b0e8-ffaddaf5a744
CSeq: 1494 BYE
Server: Asterisk PBX 18.9.0
Content-Length: 0
You can see Asterisk sets the channel to hold but then once it is transferred over (SIP REFER, accepted) the channel continues to remain on hold. The result is the destination party receiving the call will continue to hear hold music.