Hello,
I am encountering an issue that I have never faced before, and I am unsure of the cause. Here is the scenario:
- Asterisk: 20.4.0
- Asterisk IP: 192.168.5.10
- IP Phone: 192.168.5.1
Problem: I am receiving calls from the telephone operator via E1 R2 through a TDMoE Gateway. The call is answered by a queue and forwarded to extension 7400 (192.168.5.1
). The IP phone answers the call, but 30 seconds later, the call is terminated by Asterisk.
The call drops in this scenario: PSTN -> open_r2 -> Asterisk -> Queue -> PJSIP/7400
The call is terminated by Asterisk:
INVITE (SDP)
│INVITE sip:7400@192.168.5.1:5060 SIP/2.0
192.168.5.10:5060 192.168.5.1:5060 │Via: SIP/2.0/UDP 192.168.5.10:5060;rport;branch=z9hG4bKPjd20b3f03-e56a-45a4-b9f6-8417f9fc8dc5
──────────┬───────── ──────────┬─────────│From: "5433350700" <sip:5433350700@192.168.1.10>;tag=83cf1fc3-de92-4af9-a3c9-55ea2ea5bee7
09:59:19.076398 │ INVITE (SDP) │ │To: <sip:7400@192.168.5.1>
+0.024243 │ ──────────────────────────> │ │Contact: <sip:asterisk@192.168.5.10:5060>
09:59:19.100641 │ 100 Trying │ │Call-ID: afafeda2-cf47-49c2-ad27-defdd753f086
+0.191745 │ <────────────────────────── │ │CSeq: 20028 INVITE
09:59:19.292386 │ 180 Ringing │ │Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
+2.180236 │ <────────────────────────── │ │Supported: 100rel, timer, replaces, norefersub, histinfo
09:59:21.472622 │ 200 OK (SDP) │ │Session-Expires: 1800
+0.000792 │ <────────────────────────── │ │Min-SE: 90
09:59:21.473414 │ ACK │ │Max-Forwards: 70
+9.435270 │ ──────────────────────────> │ │User-Agent: Asterisk PBX 20.4.0
09:59:30.908684 │ BYE │ │Content-Type: application/sdp
+0.008793 │ ──────────────────────────> │ │Content-Length: 237
09:59:30.917477 │ 200 OK │ │
│ <────────────────────────── │ │v=0
│ │ │o=- 1564605616 1564605616 IN IP4 192.168.5.10
│ │ │s=Asterisk
│ │ │c=IN IP4 192.168.5.10
│ │ │t=0 0
│ │ │m=audio 13548 RTP/AVP 0 101
│ │ │a=rtpmap:0 PCMU/8000
│ │ │a=rtpmap:101 telephone-event/8000
│ │ │a=fmtp:101 0-16
│ │ │a=ptime:20
│ │ │a=maxptime:150
│ │ │a=sendrecv
2024/04/23 09:59:19.100641 192.168.5.1:5060 -> 192.168.5.10:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.10:5060;rport=5060;branch=z9hG4bKPjd20b3f03-e56a-45a4-b9f6-8417f9fc8dc5
From: "5433350700" <sip:5433350700@192.168.1.10>;tag=83cf1fc3-de92-4af9-a3c9-55ea2ea5bee7
To: <sip:7400@192.168.5.1>
Call-ID: afafeda2-cf47-49c2-ad27-defdd753f086
CSeq: 20028 INVITE
User-Agent: Yealink SIP-T48S 66.86.0.160
Content-Length: 0
2024/04/23 09:59:19.292386 192.168.5.1:5060 -> 192.168.5.10:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.5.10:5060;rport=5060;branch=z9hG4bKPjd20b3f03-e56a-45a4-b9f6-8417f9fc8dc5
From: "5433350700" <sip:5433350700@192.168.1.10>;tag=83cf1fc3-de92-4af9-a3c9-55ea2ea5bee7
To: <sip:7400@192.168.5.1>;tag=663955336
Call-ID: afafeda2-cf47-49c2-ad27-defdd753f086
CSeq: 20028 INVITE
Contact: <sip:7400@192.168.5.1:5060>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T48S 66.86.0.160
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0
200 OK (SDP)
│SIP/2.0 200 OK
192.168.5.10:5060 192.168.5.1:5060 │Via: SIP/2.0/UDP 192.168.5.10:5060;rport=5060;branch=z9hG4bKPjd20b3f03-e56a-45a4-b9f6-8417f9fc8dc5
──────────┬───────── ──────────┬─────────│From: "5433350700" <sip:5433350700@192.168.1.10>;tag=83cf1fc3-de92-4af9-a3c9-55ea2ea5bee7
09:59:19.076398 │ INVITE (SDP) │ │To: <sip:7400@192.168.5.1>;tag=663955336
+0.024243 │ ──────────────────────────> │ │Call-ID: afafeda2-cf47-49c2-ad27-defdd753f086
09:59:19.100641 │ 100 Trying │ │CSeq: 20028 INVITE
+0.191745 │ <────────────────────────── │ │Contact: <sip:7400@192.168.5.1:5060>
09:59:19.292386 │ 180 Ringing │ │Content-Type: application/sdp
+2.180236 │ <────────────────────────── │ │Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
09:59:21.472622 │ 200 OK (SDP) │ │User-Agent: Yealink SIP-T48S 66.86.0.160
+0.000792 │ <────────────────────────── │ │Allow-Events: talk,hold,conference,refer,check-sync
09:59:21.473414 │ ACK │ │Supported: replaces
+9.435270 │ ──────────────────────────> │ │Content-Length: 209
09:59:30.908684 │ BYE │ │
+0.008793 │ ──────────────────────────> │ │v=0
09:59:30.917477 │ 200 OK │ │o=- 20032 20032 IN IP4 192.168.5.1
│ <────────────────────────── │ │s=SDP data
│ │ │c=IN IP4 192.168.5.1
│ │ │t=0 0
│ │ │m=audio 12232 RTP/AVP 0 101
│ │ │a=rtpmap:0 PCMU/8000
│ │ │a=ptime:20
│ │ │a=rtpmap:101 telephone-event/8000
│ │ │a=fmtp:101 0-15
│ │ │a=sendrecv
ACK
│ACK sip:7400@192.168.5.1:5060 SIP/2.0
192.168.5.10:5060 192.168.5.1:5060 │Via: SIP/2.0/UDP 192.168.5.10:5060;rport;branch=z9hG4bKPja43dc6f4-ea90-479f-acdb-26662e325b8e
──────────┬───────── ──────────┬─────────│From: "5433350700" <sip:5433350700@192.168.1.10>;tag=83cf1fc3-de92-4af9-a3c9-55ea2ea5bee7
09:59:19.076398 │ INVITE (SDP) │ │To: <sip:7400@192.168.5.1>;tag=663955336
+0.024243 │ ──────────────────────────> │ │Call-ID: afafeda2-cf47-49c2-ad27-defdd753f086
09:59:19.100641 │ 100 Trying │ │CSeq: 20028 ACK
+0.191745 │ <────────────────────────── │ │Max-Forwards: 70
09:59:19.292386 │ 180 Ringing │ │User-Agent: Asterisk PBX 20.4.0
+2.180236 │ <────────────────────────── │ │Content-Length: 0
09:59:21.472622 │ 200 OK (SDP) │ │
+0.000792 │ <────────────────────────── │ │
09:59:21.473414 │ ACK │ │
+9.435270 │ ──────────────────────────> │ │
09:59:30.908684 │ BYE │ │
+0.008793 │ ──────────────────────────> │ │
09:59:30.917477 │ 200 OK │ │
│ <────────────────────────── │ │
BYE
│BYE sip:7400@192.168.5.1:5060 SIP/2.0
192.168.5.10:5060 192.168.5.1:5060 │Via: SIP/2.0/UDP 192.168.5.10:5060;rport;branch=z9hG4bKPjbb0c374d-75c7-4699-8f28-7cddc11e45ee
──────────┬───────── ──────────┬─────────│From: "5433350700" <sip:5433350700@192.168.1.10>;tag=83cf1fc3-de92-4af9-a3c9-55ea2ea5bee7
09:59:19.076398 │ INVITE (SDP) │ │To: <sip:7400@192.168.5.1>;tag=663955336
+0.024243 │ ──────────────────────────> │ │Call-ID: afafeda2-cf47-49c2-ad27-defdd753f086
09:59:19.100641 │ 100 Trying │ │CSeq: 20029 BYE
+0.191745 │ <────────────────────────── │ │Reason: Q.850;cause=16
09:59:19.292386 │ 180 Ringing │ │Max-Forwards: 70
+2.180236 │ <────────────────────────── │ │User-Agent: Asterisk PBX 20.4.0
09:59:21.472622 │ 200 OK (SDP) │ │Content-Length: 0
+0.000792 │ <────────────────────────── │ │
09:59:21.473414 │ ACK │ │
+9.435270 │ ──────────────────────────> │ │
09:59:30.908684 │ BYE │ │
+0.008793 │ ──────────────────────────> │ │
09:59:30.917477 │ 200 OK │ │
│ <────────────────────────── │ │
200 OK
2024/04/23 09:59:30.917477 192.168.5.1:5060 -> 192.168.5.10:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.10:5060;rport=5060;branch=z9hG4bKPjbb0c374d-75c7-4699-8f28-7cddc11e45ee
From: "5433350700" <sip:5433350700@192.168.1.10>;tag=83cf1fc3-de92-4af9-a3c9-55ea2ea5bee7
To: <sip:7400@192.168.5.1>;tag=663955336
Call-ID: afafeda2-cf47-49c2-ad27-defdd753f086
CSeq: 20029 BYE
User-Agent: Yealink SIP-T48S 66.86.0.160
Content-Length: 0
If I direct the call straight to the extension PJSIP/7400
instead of sending it to the queue, the call does not drop. The problem only occurs when the queue answers and forwards the call to the extension.
In this scenario, the call does not drop: PSTN -> open_r2 -> Asterisk -> PJSIP/7400
I have no idea why this is happening. I need the calls to be sent to the queue before going to the phone. I have the same setup with several other clients, but this is the only one presenting this issue, and I have no idea what is happening.
Can anyone help me?