Rtp is changed when call

below is cli: the question is why after dial the dialplan imediatly play on hold music…why not to play the early media .Hope for answer…

-- Executing [s@macro-dialGateway:1] Set("SIP/8761026-001226f5", "CALLERID(num)=83160194") in new stack
-- Executing [s@macro-dialGateway:2] Dial("SIP/8761026-001226f5", "SIP/13012345678@IBAC51,60,FgL(18000000:61000)") in new stack
   > Limit Data for this call:
   > timelimit      = 18000000 ms (18000.000 s)
   > play_warning   = 61000 ms (61.000 s)
   > play_to_caller = yes
   > play_to_callee = no
   > warning_freq   = 0 ms (0.000 s)
   > start_sound    = 
   > warning_sound  = timeleft
   > end_sound      = 

== Using SIP RTP CoS mark 5
– Called SIP/13012345678@IBAC51

0x7fe0dc20eed0 – Probation passed - setting RTP source address to 192.168.122.72:48344
– Call on SIP/IBAC51-001226f6 placed on hold
– Started music on hold, class ‘default’, on SIP/8761026-001226f5
– SIP/IBAC51-001226f6 is making progress passing it to SIP/8761026-001226f5
0x7fe0dc20eed0 – Probation passed - setting RTP source address to 192.168.122.72:48344
0x7fe17da26ee0 – Probation passed - setting RTP source address to 211.166.195.74:4020
0x7fe0dc20eed0 – Probation passed - setting RTP source address to 192.168.122.72:48344
– Stopped music on hold on SIP/8761026-001226f5
== Spawn extension (macro-dialGateway, s, 3) exited non-zero on ‘SIP/8761026-001226f5’ in macro ‘dialGateway’
== Spawn extension (Gdfs, 13012345678, 5) exited non-zero on ‘SIP/8761026-001226f5’

the sip is: plz tell me how to redirect the media to the client.


The answer almost certainly lies in the message body, which you haven’t provided.

Also note that there is a strong preference for presenting protocol logging as text from the Asterisk protocol logs, not as Wireshark images.

sorry,this is sip debug info:

<--- SIP read from UDP:223.166.194.50:10081 --->
INVITE sip:17602181234@www.call123.net SIP/2.0
Via: SIP/2.0/UDP 223.166.194.50:10081;rport;branch=z9hG4bKPj8cbcfd3382df42a0a4a788fcc59d7578
Max-Forwards: 70
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>
Contact: <sip:8761026@223.166.194.50:10081;ob>
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11104 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.20.1
Content-Type: application/sdp
Content-Length: 416

v=0
o=- 3813552801 3813552801 IN IP4 223.166.194.50
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 18 96 101
c=IN IP4 223.166.194.50
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.6.195
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1801729379 cname:32bd51d535324c1e
<------------->
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: --- (15 headers 19 lines) ---
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: Sending to 223.166.194.50:10081 (NAT)
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Sending to 223.166.194.50:10081 (NAT)
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Using INVITE request as basis request - d58bae99a0d540eda106c8de9b4ea31d
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found peer '8761026' for '8761026' from 223.166.194.50:10081
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: 
<--- Reliably Transmitting (NAT) to 223.166.194.50:10081 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 223.166.194.50:10081;branch=z9hG4bKPj8cbcfd3382df42a0a4a788fcc59d7578;received=223.166.194.50;rport=10081
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>;tag=as694a79c9
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11104 INVITE
Server: vcloud
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="vidanetwork.net", nonce="33cb87b6"
Content-Length: 0


<------------>
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Scheduling destruction of SIP dialog 'd58bae99a0d540eda106c8de9b4ea31d' in 6400 ms (Method: INVITE)
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: Really destroying SIP dialog '1567077313-18862-168417@BJC.BGI.B.DF' Method: OPTIONS
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: 
<--- SIP read from UDP:192.168.122.8:5060 --->
OPTIONS sip:192.168.141.245:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.122.8:5060;branch=z9hG4bKdovho395hzh46a9d699z4atgo;Role=3;Hpt=8e48_16;pth=0;X-HwDim=4
Call-ID: 455et46ebvd5c4v4gce6ttg3bbe5z3oe@8.122.168.192
From: <sip:SBC@192.168.122.8>;tag=vc993t5d
To: <sip:192.168.141.245>
CSeq: 1 OPTIONS
Contact: <sip:192.168.122.8:36424;transport=udp;Hpt=8e48_16>;expires=65535
Accept: application/sdp
Max-Forwards: 70
Content-Length: 0

<------------->
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: --- (10 headers 0 lines) ---
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: Sending to 192.168.122.8:5060 (NAT)
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: Looking for s in default (domain 192.168.141.245)
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: 
<--- Transmitting (NAT) to 192.168.122.8:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.122.8:5060;branch=z9hG4bKdovho395hzh46a9d699z4atgo;Role=3;Hpt=8e48_16;pth=0;X-HwDim=4;received=192.168.122.8;rport=5060
From: <sip:SBC@192.168.122.8>;tag=vc993t5d
To: <sip:192.168.141.245>;tag=as61b1ed33
Call-ID: 455et46ebvd5c4v4gce6ttg3bbe5z3oe@8.122.168.192
CSeq: 1 OPTIONS
Server: vcloud
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:192.168.141.245:5060>
Accept: application/sdp
Content-Length: 0

[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: 
<--- SIP read from UDP:223.166.194.50:10081 --->
ACK sip:17602181234@www.call123.net SIP/2.0
Via: SIP/2.0/UDP 223.166.194.50:10081;rport;branch=z9hG4bKPj8cbcfd3382df42a0a4a788fcc59d7578
Max-Forwards: 70
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>;tag=as694a79c9
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11104 ACK
Content-Length: 0

<------------->
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: --- (8 headers 0 lines) ---
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: 
<--- SIP read from UDP:223.166.194.50:10081 --->
INVITE sip:17602181234@www.call123.net SIP/2.0
Via: SIP/2.0/UDP 223.166.194.50:10081;rport;branch=z9hG4bKPjec7664c1252b4ad9af3b22c14b7ad812
Max-Forwards: 70
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>
Contact: <sip:8761026@223.166.194.50:10081;ob>
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11105 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.20.1
Authorization: Digest username="8761026", realm="vidanetwork.net", nonce="33cb87b6", uri="sip:17602181234@www.call123.net", response="c72244e8e254a28277eeaf4b89b9c135", algorithm=MD5
Content-Type: application/sdp
Content-Length: 416

v=0
o=- 3813552801 3813552801 IN IP4 223.166.194.50
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 8 0 18 96 101
c=IN IP4 223.166.194.50
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.6.195
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1801729379 cname:32bd51d535324c1e
<------------->
[Nov  5 08:13:18] VERBOSE[19440] chan_sip.c: --- (16 headers 19 lines) ---
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Sending to 223.166.194.50:10081 (NAT)
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Using INVITE request as basis request - d58bae99a0d540eda106c8de9b4ea31d
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found peer '8761026' for '8761026' from 223.166.194.50:10081
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] netsock2.c:   == Using SIP RTP CoS mark 5
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found RTP audio format 8
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found RTP audio format 0
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found RTP audio format 18
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found RTP audio format 96
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found RTP audio format 101
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found audio description format PCMA for ID 8
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found audio description format PCMU for ID 0
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found audio description format G729 for ID 18
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found audio description format iLBC for ID 96
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Peer audio RTP is at port 223.166.194.50:4000
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: Looking for 17602181234 in Gdfs (domain www.call123.net)
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: list_route: hop: <sip:8761026@223.166.194.50:10081;ob>
[Nov  5 08:13:18] VERBOSE[19440][C-00000580] chan_sip.c: 
<--- Transmitting (NAT) to 223.166.194.50:10081 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 223.166.194.50:10081;branch=z9hG4bKPjec7664c1252b4ad9af3b22c14b7ad812;received=223.166.194.50;rport=10081
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11105 INVITE
Server: vcloud
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:17602181234@14.215.46.42:5060>
Content-Length: 0


<------------>
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] pbx.c:     -- Executing [17602181234@Gdfs:1] Set("SIP/8761026-00000069", "CALLERID(num)=87184444") in new stack
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] pbx.c:     -- Executing [17602181234@Gdfs:2] Progress("SIP/8761026-00000069", "5000") in new stack
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Audio is at 16388
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: 
<--- Transmitting (NAT) to 223.166.194.50:10081 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 223.166.194.50:10081;branch=z9hG4bKPjec7664c1252b4ad9af3b22c14b7ad812;received=223.166.194.50;rport=10081
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>;tag=as4a76d290
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11105 INVITE
Server: vcloud
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: <sip:17602181234@14.215.46.42:5060>
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1551028103 1551028103 IN IP4 14.215.46.42
s=Asterisk PBX certified/11.6-cert16
c=IN IP4 14.215.46.42
t=0 0
m=audio 16388 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] pbx.c:     -- Executing [17602181234@Gdfs:3] Dial("SIP/8761026-00000069", "SIP/17602181234@HZ1,10,oghI") in new stack
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] netsock2.c:   == Using SIP RTP CoS mark 5
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Audio is at 13736
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Nov  5 08:13:18] VERBOSE[73613][C-00000580] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.8:5060:
INVITE sip:17602181234@192.168.125.8:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.141.245:5060;branch=z9hG4bK74ec64ad;rport
Max-Forwards: 70
From: "ttt sss" <sip:87184444@192.168.141.245>;tag=as3b60c381
To: <sip:17602181234@192.168.125.8:5060>
Contact: <sip:87184444@192.168.141.245:5060>
Call-ID: 76548316624918bb3e4226544176a718@192.168.141.245:5060
CSeq: 102 INVITE
User-Agent: vcloud
Date: Thu, 05 Nov 2020 00:13:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Type: application/sdp
Content-Length: 224

v=0
o=root 1453149471 1453149471 IN IP4 192.168.141.245
s=Asterisk PBX certified/11.6-cert16
c=IN IP4 192.168.141.245
t=0 0
m=audio 13736 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

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

[Nov 5 08:13:19] VERBOSE[19440] chan_sip.c:
<— SIP read from UDP:192.168.125.8:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.141.245:5060;branch=z9hG4bK74ec64ad;rport=5060
Record-Route: sip:192.168.125.8:5060;transport=udp;lr;Hpt=8e58_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=17086
Call-ID: 76548316624918bb3e4226544176a718@192.168.141.245:5060
From: "ttt sss"sip:87184444@192.168.141.245;tag=as3b60c381
To: sip:17602181234@192.168.125.8:5060;tag=21455r5b-CC-1003
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: sip:192.168.125.8:5060;transport=udp;Hpt=8e58_16;CxtId=3;TRC=ffffffff-ffffffff
P-Charging-Vector: icid-value=xgcf–20201105081318-100306917;orig-ioi=hz.charged.gd.ctcims.cn;icid-generated-at=5.1.54.15
Content-Length: 165
Content-Type: application/sdp

v=0
o=- 21938647 21938647 IN IP4 192.168.125.72
s=SBC call
c=IN IP4 192.168.125.72
t=0 0
m=audio 10098 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
<------------->
[Nov 5 08:13:19] VERBOSE[19440] chan_sip.c: — (12 headers 9 lines) —
[Nov 5 08:13:19] VERBOSE[19440][C-00000580] chan_sip.c: list_route: hop: sip:192.168.125.8:5060;transport=udp;lr;Hpt=8e58_116;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=17086
[Nov 5 08:13:19] VERBOSE[19440][C-00000580] chan_sip.c: Found RTP audio format 8
[Nov 5 08:13:19] VERBOSE[19440][C-00000580] chan_sip.c: Found audio description format PCMA for ID 8
[Nov 5 08:13:19] VERBOSE[19440][C-00000580] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Nov 5 08:13:19] VERBOSE[19440][C-00000580] chan_sip.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Nov 5 08:13:19] VERBOSE[19440][C-00000580] chan_sip.c: Peer audio RTP is at port 192.168.125.72:10098
[Nov 5 08:13:19] VERBOSE[19440] chan_sip.c: Really destroying SIP dialog ‘1478942126-32915-3022516@BJC.BGI.A.C’ Method: OPTIONS
[Nov 5 08:13:19] VERBOSE[19440] chan_sip.c: Really destroying SIP dialog ‘1932143572-48169-993648@BJC.BGI.B.CJ’ Method: OPTIONS
[Nov 5 08:13:19] VERBOSE[73613][C-00000580] app_dial.c: – Call on SIP/HZ1-0000006a placed on hold
[Nov 5 08:13:19] VERBOSE[73613][C-00000580] res_musiconhold.c: – Started music on hold, class ‘default’, on SIP/8761026-00000069
[Nov 5 08:13:19] VERBOSE[73613][C-00000580] app_dial.c: – SIP/HZ1-0000006a is making progress passing it to SIP/8761026-00000069
[Nov 5 08:13:19] VERBOSE[73613][C-00000580] res_rtp_asterisk.c: > 0x7f2898015850 – Probation passed - setting RTP source address to 192.168.125.72:10098
[Nov 5 08:13:21] VERBOSE[19440] chan_sip.c:
<— SIP read from UDP:223.166.194.50:10081 —>
CANCEL sip:17602181234@www.call123.net SIP/2.0
Via: SIP/2.0/UDP 223.166.194.50:10081;rport;branch=z9hG4bKPjec7664c1252b4ad9af3b22c14b7ad812
Max-Forwards: 70
From: sip:8761026@www.call123.net;tag=2c89757624e046a3a08d3a4456e49df6
To: sip:17602181234@www.call123.net
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11105 CANCEL
User-Agent: MicroSIP/3.20.1
Content-Length: 0

<------------->
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: --- (9 headers 0 lines) ---
[Nov  5 08:13:21] VERBOSE[19440][C-00000580] chan_sip.c: Sending to 223.166.194.50:10081 (NAT)
[Nov  5 08:13:21] VERBOSE[19440][C-00000580] chan_sip.c: 
<--- Reliably Transmitting (NAT) to 223.166.194.50:10081 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 223.166.194.50:10081;branch=z9hG4bKPjec7664c1252b4ad9af3b22c14b7ad812;received=223.166.194.50;rport=10081
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>;tag=as4a76d290
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11105 INVITE
Server: vcloud
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
[Nov  5 08:13:21] VERBOSE[19440][C-00000580] chan_sip.c: 
<--- Transmitting (NAT) to 223.166.194.50:10081 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 223.166.194.50:10081;branch=z9hG4bKPjec7664c1252b4ad9af3b22c14b7ad812;received=223.166.194.50;rport=10081
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>;tag=as4a76d290
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11105 CANCEL
Server: vcloud
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0


<------------>
[Nov  5 08:13:21] VERBOSE[73613][C-00000580] res_musiconhold.c:     -- Stopped music on hold on SIP/8761026-00000069
[Nov  5 08:13:21] VERBOSE[73613][C-00000580] chan_sip.c: Scheduling destruction of SIP dialog '76548316624918bb3e4226544176a718@192.168.141.245:5060' in 6400 ms (Method: INVITE)
[Nov  5 08:13:21] VERBOSE[73613][C-00000580] chan_sip.c: Reliably Transmitting (NAT) to 192.168.125.8:5060:
CANCEL sip:17602181234@192.168.125.8:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.141.245:5060;branch=z9hG4bK74ec64ad;rport
Max-Forwards: 70
From: "ttt sss" <sip:87184444@192.168.141.245>;tag=as3b60c381
To: <sip:17602181234@192.168.125.8:5060>
Call-ID: 76548316624918bb3e4226544176a718@192.168.141.245:5060
CSeq: 102 CANCEL
User-Agent: vcloud
Content-Length: 0


---
[Nov  5 08:13:21] VERBOSE[73613][C-00000580] chan_sip.c: Scheduling destruction of SIP dialog '76548316624918bb3e4226544176a718@192.168.141.245:5060' in 6400 ms (Method: INVITE)
[Nov  5 08:13:21] VERBOSE[73613][C-00000580] pbx.c:   == Spawn extension (Gdfs, 17602181234, 3) exited non-zero on 'SIP/8761026-00000069'
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: Really destroying SIP dialog '1410331960-56747-43961@BJC.BGI.B.HG' Method: OPTIONS
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: 
<--- SIP read from UDP:192.168.125.8:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.141.245:5060;branch=z9hG4bK74ec64ad;rport=5060
Call-ID: 76548316624918bb3e4226544176a718@192.168.141.245:5060
From: "ttt sss"<sip:87184444@192.168.141.245>;tag=as3b60c381
To: <sip:17602181234@192.168.125.8:5060>;tag=le49ra41
CSeq: 102 CANCEL
Content-Length: 0

<------------->
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: --- (7 headers 0 lines) ---
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: 
<--- SIP read from UDP:192.168.125.8:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.141.245:5060;branch=z9hG4bK74ec64ad;rport=5060
Call-ID: 76548316624918bb3e4226544176a718@192.168.141.245:5060
From: "ttt sss"<sip:87184444@192.168.141.245>;tag=as3b60c381
To: <sip:17602181234@192.168.125.8:5060>;tag=21455r5b-CC-1003
CSeq: 102 INVITE
Warning: 399 192.168.125.8 "SS250200F156L921[00000] Cancel received on initial invite"
Content-Length: 0

<------------->
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: --- (8 headers 0 lines) ---
[Nov  5 08:13:21] VERBOSE[19440][C-00000580] chan_sip.c: Transmitting (NAT) to 192.168.125.8:5060:
ACK sip:192.168.125.8:5060;transport=udp;Hpt=8e58_16;CxtId=3;TRC=ffffffff-ffffffff SIP/2.0
Via: SIP/2.0/UDP 192.168.141.245:5060;branch=z9hG4bK74ec64ad;rport
Max-Forwards: 70
From: "ttt sss" <sip:87184444@192.168.141.245>;tag=as3b60c381
To: <sip:17602181234@192.168.125.8:5060>;tag=21455r5b-CC-1003
Contact: <sip:87184444@192.168.141.245:5060>
Call-ID: 76548316624918bb3e4226544176a718@192.168.141.245:5060
CSeq: 102 ACK
User-Agent: vcloud
Content-Length: 0


---
[Nov  5 08:13:21] VERBOSE[19440][C-00000580] chan_sip.c: Scheduling destruction of SIP dialog '76548316624918bb3e4226544176a718@192.168.141.245:5060' in 6400 ms (Method: INVITE)
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: 
<--- SIP read from UDP:113.81.211.185:32042 --->

<------------->
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: Really destroying SIP dialog '1933521084-38613-148217@BJC.BGI.G.BBD' Method: OPTIONS
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: Really destroying SIP dialog '1266349544-55719-851760@BJC.BGI.B.BJ' Method: OPTIONS
[Nov  5 08:13:21] VERBOSE[19440] chan_sip.c: 
<--- SIP read from UDP:223.166.194.50:10081 --->
ACK sip:17602181234@www.call123.net SIP/2.0
Via: SIP/2.0/UDP 223.166.194.50:10081;rport;branch=z9hG4bKPjec7664c1252b4ad9af3b22c14b7ad812
Max-Forwards: 70
From: <sip:8761026@www.call123.net>;tag=2c89757624e046a3a08d3a4456e49df6
To: <sip:17602181234@www.call123.net>;tag=as4a76d290
Call-ID: d58bae99a0d540eda106c8de9b4ea31d
CSeq: 11105 ACK
Content-Length: 0

The endpoint at 192.168.125.8 provided an SDP with “sendonly” in its 183 Session Progress which chan_sip interpreted as being put on hold.

To expand on that, although sendonly technically means that the device is going to source media but not accept it, the reality is that it is the standard way that phones indicate on hold, even though they don’t source music on hold themselves, and, as it is normally phones, not other switches, that indicate hold, Asterisk treats it as “go on hold”, and starts generating its own music on hold.