HI,
I am new to Asterisk code. I am currently working on a 3-way conference call. For that, if we send a subscriber message, it shows “event conference not present”, and a notify message is coming as a 400 bad request from Asterisk.
I need to know what are the events accepted by asterisk and what event we need to send to get 200 ok for subscriber message?
There is no support for any “conference” related event type, if that’s what you are actually wanting.
Hi,
I have one doubt. If we are not sending any subscribe message to Asterisk, and we are sending a REFER and NOTIFY, we are receiving a “202 Accepted” for the REFER, but for the NOTIFY we are getting a “400 Bad Request”. May I know the reason for this?
Thank you
You’d need to provide the SIP trace to show the actual signaling.
That shows the first line of the SIP requests. It does not show the contents. A SIP trace for PJSIP can be done in Asterisk using the “pjsip set logger on” CLI command.
This is SIP trace in Asterisk
Asterisk Ready.
– Remote UNIX connection
– Remote UNIX connection disconnected
– Remote UNIX connection
<— Received SIP request (2983 bytes) from UDP:172.22.0.100:5060 —>
INVITEsip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org SIP/2.0
Record-Route: sip:mo@172.22.0.20:6060;r2=on;lr=on;ftag=d126e25c;did=e09.c2d2
Record-Route: sip:mo@172.22.0.20:6060;transport=tcp;r2=on;lr=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mtas.ims.mnc001.mcc001.3gppnetwork.org;call=orig;lr, sip:iscmark@scscf.ims.mnc001.mcc001.3gppnetwork.org;lr;s=2;h=0;d=0;a=7369703a3930303030313540696d732e6d6e633030312e6d63633030312e336770706e6574776f726b2e6f7267
Record-Route: sip:mo@172.22.0.21;r2=on;lr=on;ftag=d126e25c;rm=8;did=e09.c5e
Record-Route: sip:mo@172.22.0.21;transport=tcp;r2=on;lr=on;ftag=d126e25c;rm=8;did=e09.c5e
Via: SIP/2.0/UDP 172.22.0.100:5060;branch=z9hG4bK549c.643dd46.0
Via: SIP/2.0/UDP 172.22.0.20:6060;rport=6060;received=172.22.0.20;branch=z9hG4bK549c.d412ba192c47defdbda3bd9333c11a2e.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK549c.788966060a4fab29db529bcb7d3a2d5a.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;received=10.102.0.7;branch=z9hG4bK-524287-1—06a15dd4b65db9d7;rport=43793;transport=TCP
Max-Forwards: 67
P-Served-User: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;sescase=orig;regstate=reg
Contact: sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2;+sip.instance=“urn:gsma:imei:35245688-577195-0”;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”;video
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 1 INVITE
Session-Expires: 1800
Accept: application/sdp, application/3gpp-ims+xml
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, INFO, REFER, NOTIFY, MESSAGE, PRACK
Content-Type: application/sdp
Supported: timer, 100rel, sec-agree
User-Agent: Samsung_SM-A166P_YJ1_IMS_6.0
Accept-Contact: *;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
P-Early-Media: supported
P-Access-Network-Info: 3GPP-NR-TDD;utran-cell-id-3gpp=00101000001008388001
Content-Length: 752
P-Charging-Vector: icid-value=495653AC1600153C0000000202000000;icid-generated-at=172.22.0.21
P-Asserted-Identity: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org
P-Visited-Network-ID: ims.mnc001.mcc001.3gppnetwork.org
v=0
o=SAMSUNG-IMS-UE 513818984327 0 IN IP4 172.22.0.16
s=SS VOIP
t=0 0
a=rtpengine:0374255ce4eb
m=audio 35498 RTP/AVP 112 110 120 96 118 111 107
c=IN IP4 172.22.0.16
b=AS:42
b=RR:0
b=RS:0
a=rtpmap:112 EVS/16000
a=fmtp:112 br=5.9-24.4;bw=nb-swb;ch-aw-recv=2;max-red=0
a=rtpmap:110 AMR-WB/16000
a=fmtp:110 octet-align=1;mode-change-capability=2;max-red=0
a=rtpmap:120 AMR-WB/16000
a=fmtp:120 mode-change-capability=2;max-red=0
a=rtpmap:96 AMR/8000
a=fmtp:96 octet-align=1;mode-change-capability=2;max-red=0
a=rtpmap:118 AMR/8000
a=fmtp:118 mode-change-capability=2;max-red=0
a=rtpmap:111 telephone-event/16000
a=fmtp:111 0-15
a=rtpmap:107 telephone-event/8000
a=fmtp:107 0-15
a=sendrecv
a=rtcp:35499
a=ptime:20
a=maxptime:240
<— Transmitting SIP response (1035 bytes) to UDP:172.22.0.100:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.22.0.100:5060;rport=5060;received=172.22.0.100;branch=z9hG4bK549c.643dd46.0
Via: SIP/2.0/UDP 172.22.0.20:6060;rport=6060;received=172.22.0.20;branch=z9hG4bK549c.d412ba192c47defdbda3bd9333c11a2e.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK549c.788966060a4fab29db529bcb7d3a2d5a.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;rport=43793;received=10.102.0.7;branch=z9hG4bK-524287-1—06a15dd4b65db9d7;transport=TCP
Record-Route: sip:mo@172.22.0.100:5060;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Record-Route: sip:mo@172.22.0.20:6060;transport=tcp;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Record-Route: sip:mo@172.22.0.21;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Record-Route: sip:mo@172.22.0.21;transport=tcp;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org
CSeq: 1 INVITE
Server: Asterisk PBX 22.7.0
Content-Length: 0
== Using SIP RTP Audio TOS bits 184
– Executing [mmtel@IMS_INC:1] NoOp(“PJSIP/ims_core-00000000”, “MMTel Conference Factory”) in new stack
– Executing [mmtel@IMS_INC:2] Answer(“PJSIP/ims_core-00000000”, “”) in new stack
> 0x77f5a402a1f0 – Strict RTP learning after remote address set to: 172.22.0.16:35498
<— Transmitting SIP response (1671 bytes) to UDP:172.22.0.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.0.100:5060;rport=5060;received=172.22.0.100;branch=z9hG4bK549c.643dd46.0
Via: SIP/2.0/UDP 172.22.0.20:6060;rport=6060;received=172.22.0.20;branch=z9hG4bK549c.d412ba192c47defdbda3bd9333c11a2e.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK549c.788966060a4fab29db529bcb7d3a2d5a.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;rport=43793;received=10.102.0.7;branch=z9hG4bK-524287-1—06a15dd4b65db9d7;transport=TCP
Record-Route: sip:mo@172.22.0.100:5060;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Record-Route: sip:mo@172.22.0.20:6060;transport=tcp;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Record-Route: sip:mo@172.22.0.21;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Record-Route: sip:mo@172.22.0.21;transport=tcp;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
CSeq: 1 INVITE
Server: Asterisk PBX 22.7.0
Contact: sip:172.22.0.105:5090
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 313
v=0
o=- 513818984327 2 IN IP4 172.22.0.105
s=Asterisk
c=IN IP4 172.22.0.105
t=0 0
m=audio 10006 RTP/AVP 112 111
a=rtpmap:112 EVS/16000
a=fmtp:112 hf-only=1;dtx-recv=0;max-red=0;br=5.9-24.4;bw=nb-swb;ch-aw-recv=-1
a=rtpmap:111 telephone-event/16000
a=fmtp:111 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
<— Received SIP request (845 bytes) from UDP:172.22.0.20:6060 —>
ACK sip:172.22.0.105:5090 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.20:6060;branch=z9hG4bK549c.cb2c30c803fcbea5f1fb0c9c5c5eecf1.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK549c.b37add4c0f09e9f99886f46b5c37a0b2.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;received=10.102.0.7;branch=z9hG4bK-524287-1—7d1785755d07ed02;rport=43793;transport=TCP
Max-Forwards: 68
Proxy-Require: sec-agree
Require: sec-agree
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 1 ACK
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, INFO, REFER, NOTIFY, MESSAGE, PRACK
Supported: sec-agree
User-Agent: Samsung_SM-A166P_YJ1_IMS_6.0
P-Preferred-Identity: tel:9000015
Content-Length: 0
<— Received SIP request (1436 bytes) from UDP:172.22.0.20:6060 —>
REFER sip:172.22.0.105:5090 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.20:6060;branch=z9hG4bK249c.471dc49002f00ed2fd49d1120773e1dd.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK249c.b715fd4f4f14feb1d7f7de9e0ff9f6f5.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;received=10.102.0.7;branch=z9hG4bK-524287-1—d79fecf992f96551;rport=43793;transport=TCP
Max-Forwards: 68
Proxy-Require: sec-agree
Require: sec-agree
Contact: sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2;+sip.instance=“urn:gsma:imei:35245688-577195-0”;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”;video
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 2 REFER
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, INFO, REFER, NOTIFY, MESSAGE, PRACK
Supported: sec-agree
User-Agent: Samsung_SM-A166P_YJ1_IMS_6.0
Refer-To: sip:9000016@ims.mnc001.mcc001.3gppnetwork.org;user=phone;method=INVITE?Replaces=6V9LMUUFbc_L8fjs_5c2KA..%4010.102.0.7%3Bto-tag%3D2aaef515%3Bfrom-tag%3D6e80047d
Referred-By: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org
Event: refer
P-Preferred-Identity: tel:9000015
Accept-Contact: *;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
P-Access-Network-Info: 3GPP-NR-TDD;utran-cell-id-3gpp=00101000001008388001
Content-Length: 0
<— Transmitting SIP response (857 bytes) to UDP:172.22.0.20:6060 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 172.22.0.20:6060;rport=6060;received=172.22.0.20;branch=z9hG4bK249c.471dc49002f00ed2fd49d1120773e1dd.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK249c.b715fd4f4f14feb1d7f7de9e0ff9f6f5.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;rport=43793;received=10.102.0.7;branch=z9hG4bK-524287-1—d79fecf992f96551;transport=TCP
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
CSeq: 2 REFER
Expires: 600
Contact: sip:172.22.0.105:5090
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 22.7.0
Content-Length: 0
<— Transmitting SIP request (992 bytes) to UDP:172.22.0.100:5060 —>
NOTIFY sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.105:5090;rport;branch=z9hG4bKPjcfdaa742-a308-427f-8c7f-a62d601729ff
From: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
To: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Contact: sip:172.22.0.105:5090
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 3200 NOTIFY
Route: sip:mo@172.22.0.100:5060;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.20:6060;transport=tcp;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.21;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Route: sip:mo@172.22.0.21;transport=tcp;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Event: refer
Subscription-State: active;expires=600
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Type: message/sipfrag;version=2.0
Content-Length: 20
SIP/2.0 100 Trying
<— Transmitting SIP request (1007 bytes) to UDP:172.22.0.100:5060 —>
NOTIFY sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.105:5090;rport;branch=z9hG4bKPjbf42c4e8-969c-4f94-94f6-a4c1068b1959
From: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
To: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Contact: sip:172.22.0.105:5090
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 3201 NOTIFY
Route: sip:mo@172.22.0.100:5060;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.20:6060;transport=tcp;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.21;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Route: sip:mo@172.22.0.21;transport=tcp;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Event: refer
Subscription-State: terminated;reason=noresource
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Type: message/sipfrag;version=2.0
Content-Length: 25
SIP/2.0 400 Bad Request
> 0x77f5a402a1f0 -- Strict RTP qualifying stream type: audio
<— Received SIP response (617 bytes) from UDP:172.22.0.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.0.105:5090;received=172.22.0.105;rport=5090;branch=z9hG4bKPjcfdaa742-a308-427f-8c7f-a62d601729ff
Contact: sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2;+sip.instance=“urn:gsma:imei:35245688-577195-0”;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”;video
To: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
From: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 3200 NOTIFY
Server: Samsung_SM-A166P_YJ1_IMS_6.0
Content-Length: 0
<— Received SIP response (617 bytes) from UDP:172.22.0.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.0.105:5090;received=172.22.0.105;rport=5090;branch=z9hG4bKPjbf42c4e8-969c-4f94-94f6-a4c1068b1959
Contact: sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2;+sip.instance=“urn:gsma:imei:35245688-577195-0”;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”;video
To: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
From: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 3201 NOTIFY
Server: Samsung_SM-A166P_YJ1_IMS_6.0
Content-Length: 0
<— Received SIP request (959 bytes) from UDP:172.22.0.20:6060 —>
BYE sip:172.22.0.105:5090 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.20:6060;branch=z9hG4bK349c.e498d3cce4f3e01cb0c78b8e2e00fed0.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK349c.003de631f9220f6042d09e3cc39fda8e.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;received=10.102.0.7;branch=z9hG4bK-524287-1—ad973634d3a4e0ad;rport=43793;transport=TCP
Max-Forwards: 68
Proxy-Require: sec-agree
Require: sec-agree
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 3 BYE
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, INFO, REFER, NOTIFY, MESSAGE, PRACK
Supported: sec-agree
User-Agent: Samsung_SM-A166P_YJ1_IMS_6.0
Reason: SIP; text=“User Disconnected”
P-Preferred-Identity: tel:9000015
P-Access-Network-Info: 3GPP-NR-TDD;utran-cell-id-3gpp=00101000001008388001
Content-Length: 0
<— Transmitting SIP response (636 bytes) to UDP:172.22.0.20:6060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.22.0.20:6060;rport=6060;received=172.22.0.20;branch=z9hG4bK349c.e498d3cce4f3e01cb0c78b8e2e00fed0.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK349c.003de631f9220f6042d09e3cc39fda8e.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;rport=43793;received=10.102‑.0.7;branch=z9hG4bK-524287-1—ad973634d3a4e0ad;transport=TCP
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
CSeq: 3 BYE
Server: Asterisk PBX 22.7.0
Content-Length: 0
== Spawn extension (IMS_INC, mmtel, 2) exited non-zero on ‘PJSIP/ims_core-00000000’
It’s an invalid/unsupported transfer scenario. The channel was not connected to anything else, so there was nothing to transfer.
Is there any way to create the channel. how it creates
You would need to Dial() something.
That doesn’t make sense to me. I suspect the real reason is that redirect_method is set to, the default, “user”, when it needs to be one of the “uri-” options.
@david551 Within the core you can’t transfer the dialplan directly somewhere, you can only transfer channels you are dialed/bridged to. I don’t know what will happen with the other PJSIP options for it.
The OP probably needs to try uri-pjsip, as that sounds like the only option that might work.
Hi
I need to know one thing
Ue is sending REFER message, for that asterisk is sending “202 Acepted” and it is getting NOTIFY message from asterisk but we are getting “400 Bad request” what does it mean
<— Received SIP request (1436 bytes) from UDP:172.22.0.20:6060 —>
REFER sip:172.22.0.105:5090 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.20:6060;branch=z9hG4bK249c.471dc49002f00ed2fd49d1120773e1dd.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK249c.b715fd4f4f14feb1d7f7de9e0ff9f6f5.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;received=10.102.0.7;branch=z9hG4bK-524287-1—d79fecf992f96551;rport=43793;transport=TCP
Max-Forwards: 68
Proxy-Require: sec-agree
Require: sec-agree
Contact: sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2;+sip.instance=“urn:gsma:imei:35245688-577195-0”;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”;video
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 2 REFER
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, INFO, REFER, NOTIFY, MESSAGE, PRACK
Supported: sec-agree
User-Agent: Samsung_SM-A166P_YJ1_IMS_6.0
Refer-To: sip:9000016@ims.mnc001.mcc001.3gppnetwork.org;user=phone;method=INVITE?Replaces=6V9LMUUFbc_L8fjs_5c2KA..%4010.102.0.7%3Bto-tag%3D2aaef515%3Bfrom-tag%3D6e80047d
Referred-By: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org
Event: refer
P-Preferred-Identity: tel:9000015
Accept-Contact: *;+g.3gpp.icsi-ref=“urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel”
P-Access-Network-Info: 3GPP-NR-TDD;utran-cell-id-3gpp=00101000001008388001
Content-Length: 0
<— Transmitting SIP response (857 bytes) to UDP:172.22.0.20:6060 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 172.22.0.20:6060;rport=6060;received=172.22.0.20;branch=z9hG4bK249c.471dc49002f00ed2fd49d1120773e1dd.0;i=1
Via: SIP/2.0/TCP 172.22.0.21;branch=z9hG4bK249c.b715fd4f4f14feb1d7f7de9e0ff9f6f5.0;i=82
Via: SIP/2.0/TCP 10.102.0.7:5060;rport=43793;received=10.102.0.7;branch=z9hG4bK-524287-1—d79fecf992f96551;transport=TCP
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
From: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
To: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
CSeq: 2 REFER
Expires: 600
Contact: sip:172.22.0.105:5090
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 22.7.0
Content-Length: 0
<— Transmitting SIP request (992 bytes) to UDP:172.22.0.100:5060 —>
NOTIFY sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.105:5090;rport;branch=z9hG4bKPjcfdaa742-a308-427f-8c7f-a62d601729ff
From: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
To: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Contact: sip:172.22.0.105:5090
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 3200 NOTIFY
Route: sip:mo@172.22.0.100:5060;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.20:6060;transport=tcp;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.21;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Route: sip:mo@172.22.0.21;transport=tcp;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Event: refer
Subscription-State: active;expires=600
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Type: message/sipfrag;version=2.0
Content-Length: 20
SIP/2.0 100 Trying
<— Transmitting SIP request (1007 bytes) to UDP:172.22.0.100:5060 —>
NOTIFY sip:9000015@10.102.0.7:5060;alias=10.102.0.7~43793~2 SIP/2.0
Via: SIP/2.0/UDP 172.22.0.105:5090;rport;branch=z9hG4bKPjbf42c4e8-969c-4f94-94f6-a4c1068b1959
From: sip:mmtel@conf-factory.ims.mnc001.mcc001.3gppnetwork.org;tag=b2971dea-f8b3-4d43-9453-2493ae0c4b27
To: sip:9000015@ims.mnc001.mcc001.3gppnetwork.org;tag=d126e25c
Contact: sip:172.22.0.105:5090
Call-ID: WP7_QsKVMbG9nwMBR7gEkg..@10.102.0.7
CSeq: 3201 NOTIFY
Route: sip:mo@172.22.0.100:5060;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.20:6060;transport=tcp;lr;r2=on;ftag=d126e25c;did=e09.c2d2
Route: sip:mo@172.22.0.21;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Route: sip:mo@172.22.0.21;transport=tcp;lr;r2=on;ftag=d126e25c;rm=8;did=e09.c5e
Event: refer
Subscription-State: terminated;reason=noresource
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Type: message/sipfrag;version=2.0
Content-Length: 25
SIP/2.0 400 Bad Request
0x77f5a402a1f0 – Strict RTP qualifying stream type: audio
so what does it mean
I already answered that question in this thread earlier.