it is a NO ANSWER where the caller gave up
So what is returning a 486? A user device or an upstream? And if the caller gave up that would be a CANCEL which is neither a 480 or 486.
what I mean by caller gave up is callee did not pick and call dropped after RINGING
upstream and user device
You are going to need to be more specific, Caller → Callee and then what is cancelling the call? Are you limiting the ringtime in the Dial() so if the Callee doesn’t pick up in 30 seconds, hangup? 480/486 are responses from the callee’s side meaning their device returned it. If the Dial() timeout that would be a completely different response.
Ringing time is over and call drops because callee did not answer the call
we’re connected to MNO so this is the case, yes
You’re going to need to show a debug of this call. Asterisk canceling the Dial() isn’t going to result in a 486 so something is missing here. We’re going to need a call that returns a 486 when Dial() cancels the call.
here is the debug
<— SIP read from UDP:197.211.58.14:37664 —>
REGISTER sip:102.129.36.92:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—d7b07ecc155e21fd;rport
Max-Forwards: 70
Contact: sip:5337260406@197.211.58.14:37664;transport=UDP;rinstance=9b4eaf23cf87c36c
To: sip:5337260406@102.129.36.92:5060;transport=UDP
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=9952f23d
Call-ID: 0ChtCRH2q9zMUi4zg1ET4Q…
CSeq: 9 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.13 v2.10.18.3
Authorization: Digest username=“5337260406”,realm=“asterisk”,nonce=“46ce6751”,uri=“sip:102.129.36.92:5060;transport=UDP”,response=“cba1f9d9899b7b16eab9a19b943a3080”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Sending to 197.211.58.14:37664 (NAT)
Sending to 197.211.58.14:37664 (NAT)
<— Transmitting (NAT) to 197.211.58.14:37664 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—d7b07ecc155e21fd;received=197.211.58.14;rport=37664
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=9952f23d
To: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=as07e4c5f7
Call-ID: 0ChtCRH2q9zMUi4zg1ET4Q…
CSeq: 9 REGISTER
Server: Asterisk PBX 13.38.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“5518163e”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0ChtCRH2q9zMUi4zg1ET4Q…’ in 32000 ms (Method: REGISTER)
<— SIP read from UDP:197.211.58.14:37664 —>
REGISTER sip:102.129.36.92:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—78c204b64d39a81a;rport
Max-Forwards: 70
Contact: sip:5337260406@197.211.58.14:37664;transport=UDP;rinstance=9b4eaf23cf87c36c
To: sip:5337260406@102.129.36.92:5060;transport=UDP
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=9952f23d
Call-ID: 0ChtCRH2q9zMUi4zg1ET4Q…
CSeq: 10 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.13 v2.10.18.3
Authorization: Digest username=“5337260406”,realm=“asterisk”,nonce=“5518163e”,uri=“sip:102.129.36.92:5060;transport=UDP”,response=“a7ab45da014e02df0dce09bd918794ae”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Sending to 197.211.58.14:37664 (NAT)
<— Transmitting (NAT) to 197.211.58.14:37664 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—78c204b64d39a81a;received=197.211.58.14;rport=37664
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=9952f23d
To: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=as07e4c5f7
Call-ID: 0ChtCRH2q9zMUi4zg1ET4Q…
CSeq: 10 REGISTER
Server: Asterisk PBX 13.38.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:5337260406@197.211.58.14:37664;transport=UDP;rinstance=9b4eaf23cf87c36c;expires=60
Date: Sat, 10 Sep 2022 16:14:07 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘0ChtCRH2q9zMUi4zg1ET4Q…’ in 32000 ms (Method: REGISTER)
mkel*CLI>
<— SIP read from UDP:197.211.58.14:37664 —>
INVITE sip:2347038284899@102.129.36.92:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—27f3ea9f09b99b2b;rport
Max-Forwards: 70
Contact: sip:5337260406@197.211.58.14:37664;transport=UDP
To: sip:2347038284899@102.129.36.92:5060
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.13 v2.10.18.3
Allow-Events: presence, kpml, talk
Content-Length: 331
v=0
o=Z 0 764905876 IN IP4 192.168.0.100
s=Z
c=IN IP4 192.168.0.100
t=0 0
m=audio 57098 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
— (13 headers 13 lines) —
Sending to 197.211.58.14:37664 (NAT)
Sending to 197.211.58.14:37664 (NAT)
Using INVITE request as basis request - 24mhrqKqSTdrLzGpONTftg…
Found peer ‘5337260406’ for ‘5337260406’ from 197.211.58.14:37664
<— Reliably Transmitting (NAT) to 197.211.58.14:37664 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—27f3ea9f09b99b2b;received=197.211.58.14;rport=37664
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
To: sip:2347038284899@102.129.36.92:5060;tag=as7a489290
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 1 INVITE
Server: Asterisk PBX 13.38.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“13af5697”
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘24mhrqKqSTdrLzGpONTftg…’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:197.211.58.14:37664 —>
ACK sip:2347038284899@102.129.36.92:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—27f3ea9f09b99b2b;rport
Max-Forwards: 70
To: sip:2347038284899@102.129.36.92:5060;tag=as7a489290
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 1 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:197.211.58.14:37664 —>
INVITE sip:2347038284899@102.129.36.92:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—eeeb5c745b03db34;rport
Max-Forwards: 70
Contact: sip:5337260406@197.211.58.14:37664;transport=UDP
To: sip:2347038284899@102.129.36.92:5060
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.13 v2.10.18.3
Authorization: Digest username=“5337260406”,realm=“asterisk”,nonce=“13af5697”,uri=“sip:2347038284899@102.129.36.92:5060;transport=UDP”,response=“35c95e20dd3aefea03623d61155cd07b”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 331
v=0
o=Z 0 764905876 IN IP4 192.168.0.100
s=Z
c=IN IP4 192.168.0.100
t=0 0
m=audio 57098 RTP/AVP 106 9 98 101 0 8 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
— (14 headers 13 lines) —
Sending to 197.211.58.14:37664 (NAT)
Using INVITE request as basis request - 24mhrqKqSTdrLzGpONTftg…
Found peer ‘5337260406’ for ‘5337260406’ from 197.211.58.14:37664
== Using SIP RTP CoS mark 5
Got SDP version 764905876 and unique parts [Z 0 IN IP4 192.168.0.100]
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 98
Found RTP audio format 101
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found audio description format opus for ID 106
Found unknown media description format telephone-event for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g729), peer - audio=(ulaw|gsm|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
0x7ff4e800bf50 – Strict RTP learning after remote address set to: 192.168.0.100:57098
Peer audio RTP is at port 192.168.0.100:57098
Looking for 2347038284899 in a2billing (domain 102.129.36.92)
sip_route_dump: route/path hop: sip:5337260406@197.211.58.14:37664;transport=UDP
<— Transmitting (NAT) to 197.211.58.14:37664 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—eeeb5c745b03db34;received=197.211.58.14;rport=37664
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
To: sip:2347038284899@102.129.36.92:5060
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 2 INVITE
Server: Asterisk PBX 13.38.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2347038284899@102.129.36.92:5060
Content-Length: 0
<------------>
– Executing [2347038284899@a2billing:1] NoOp(“SIP/5337260406-00000004”, “a2billing start”) in new stack
– Executing [2347038284899@a2billing:2] Set(“SIP/5337260406-00000004”, “A2BACCOUNTCODE=5337260406”) in new stack
– Executing [2347038284899@a2billing:3] MYSQL(“SIP/5337260406-00000004”, “Connect CONNID localhost a2billinguser a2billing mya2billing”) in new stack
– Executing [2347038284899@a2billing:4] MYSQL(“SIP/5337260406-00000004”, “Query RESULTID 1 SELECT max_concurrent
FROM cc_card
WHERE username
= 5337260406”) in new stack
– Executing [2347038284899@a2billing:5] GotoIf(“SIP/5337260406-00000004”, “0?lbl_a2billing_0:”) in new stack
– Executing [2347038284899@a2billing:6] MYSQL(“SIP/5337260406-00000004”, “Fetch vdp_tmp 2 MAXCHANNELS”) in new stack
– Executing [2347038284899@a2billing:7] GotoIf(“SIP/5337260406-00000004”, “0?lbl_a2billing_0:”) in new stack
– Executing [2347038284899@a2billing:8] MYSQL(“SIP/5337260406-00000004”, “Clear 2”) in new stack
– Executing [2347038284899@a2billing:9] MYSQL(“SIP/5337260406-00000004”, “Disconnect 1”) in new stack
– Executing [2347038284899@a2billing:10] NoOp(“SIP/5337260406-00000004”, “Maximum Channels Allowed = 10”) in new stack
– Executing [2347038284899@a2billing:11] NoOp(“SIP/5337260406-00000004”, “Channels in use = 0”) in new stack
– Executing [2347038284899@a2billing:12] Set(“SIP/5337260406-00000004”, “CURRENTCHANNELS=0”) in new stack
– Executing [2347038284899@a2billing:13] Set(“SIP/5337260406-00000004”, “REMAININGCHANNELS=10.000000”) in new stack
– Executing [2347038284899@a2billing:14] NoOp(“SIP/5337260406-00000004”, “Remaining channels available = 10.000000”) in new stack
– Executing [2347038284899@a2billing:15] Set(“SIP/5337260406-00000004”, “GROUP(OUT)=5337260406”) in new stack
– Executing [2347038284899@a2billing:16] GotoIf(“SIP/5337260406-00000004”, “1?:lbl_a2billing_1”) in new stack
– Executing [2347038284899@a2billing:17] NoOp(“SIP/5337260406-00000004”, “Call Allowed as channels available”) in new stack
– Executing [2347038284899@a2billing:18] AGI(“SIP/5337260406-00000004”, “a2billing.php,2”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
– AGI Script Executing Application: (DIAL) Options: (SIP/mtn/+2347038284899,60,iL(19963000:61000:30000))
Limit Data for this call:
timelimit = 19963000 ms (19963.000 s)
play_warning = 61000 ms (61.000 s)
play_to_caller = yes
play_to_callee = no
warning_freq = 30000 ms (30.000 s)
start_sound =
warning_sound = timeleft
end_sound =
== Using SIP RTP CoS mark 5
Audio is at 19120
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 197.210.19.91:5060:
INVITE sip:+2347038284899@197.210.19.91 SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK3ca1351f;rport
Max-Forwards: 70
From: sip:+447749931908@102.129.36.92;tag=as513b94b1
To: sip:+2347038284899@197.210.19.91
Contact: sip:+447749931908@102.129.36.92:5060
Call-ID: 0ad7b42a0945df613657e58c6375f0b9@102.129.36.92:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.38.3
Date: Sat, 10 Sep 2022 16:14:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: “+447749931908” sip:+447749931908@102.129.36.92
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 561962203 561962203 IN IP4 102.129.36.92
s=Asterisk PBX 13.38.3
c=IN IP4 102.129.36.92
t=0 0
m=audio 19120 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
– Called SIP/mtn/+2347038284899
<— SIP read from UDP:197.210.19.91:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK3ca1351f;rport=5060
Call-ID: 0ad7b42a0945df613657e58c6375f0b9@102.129.36.92:5060
From: sip:+447749931908@102.129.36.92;tag=as513b94b1
To: sip:+2347038284899@197.210.19.91
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Reliably Transmitting (NAT) to 197.210.19.91:5060:
OPTIONS sip:197.210.19.91 SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK6f4bc342;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@102.129.36.92;tag=as36e9dc5b
Contact: sip:asterisk@102.129.36.92:5060
Call-ID: 1137544836fb254c20e2f8ea504151fd@102.129.36.92:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.38.3
Date: Sat, 10 Sep 2022 16:14:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:197.210.19.91:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK6f4bc342;rport=5060
Call-ID: 1137544836fb254c20e2f8ea504151fd@102.129.36.92:5060
From: "asterisk"sip:asterisk@102.129.36.92;tag=as36e9dc5b
To: sip:197.210.19.91;tag=01122773567416
CSeq: 102 OPTIONS
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Supported: 100rel,resource-priority,precondition
Accept-Resource-Priority: q735.0,q735.1,q735.2,q735.3,q735.4
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘1137544836fb254c20e2f8ea504151fd@102.129.36.92:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 105.112.159.104:5060:
OPTIONS sip:105.112.159.104;user=phone SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK17701983;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@102.129.36.92;tag=as3c7a022f
To: sip:105.112.159.104;user=phone
Contact: sip:asterisk@102.129.36.92:5060
Call-ID: 3e67ed23031f108621fba3670728d552@102.129.36.92:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.38.3
Date: Sat, 10 Sep 2022 16:14:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:105.112.159.104:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK17701983;rport=5060
Call-ID: 3e67ed23031f108621fba3670728d552@102.129.36.92:5060
From: "asterisk"sip:asterisk@102.129.36.92;tag=as3c7a022f
To: sip:105.112.159.104;user=phone;tag=9aa86cs8
CSeq: 102 OPTIONS
Allow: OPTIONS,NOTIFY,SUBSCRIBE,INFO,REGISTER,MESSAGE,REFER,UPDATE,PRACK,BYE,CANCEL,ACK,INVITE
Supported: privacy,precondition,100rel
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘3e67ed23031f108621fba3670728d552@102.129.36.92:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 194.28.167.186:5060:
OPTIONS sip:194.28.167.186 SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK6536f1dd;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@102.129.36.92;tag=as4a252aff
Contact: sip:asterisk@102.129.36.92:5060
Call-ID: 1933ffc5788bff4b32a949371708afed@102.129.36.92:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.38.3
Date: Sat, 10 Sep 2022 16:14:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Reliably Transmitting (NAT) to 194.28.167.31:5060:
OPTIONS sip:194.28.167.31 SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK73c76c4a;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@102.129.36.92;tag=as5f685ee4
Contact: sip:asterisk@102.129.36.92:5060
Call-ID: 0da1ce7002aa7b302f989a0d420e7161@102.129.36.92:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.38.3
Date: Sat, 10 Sep 2022 16:14:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
Reliably Transmitting (NAT) to 194.28.167.32:5060:
OPTIONS sip:194.28.167.32 SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK7f229723;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@102.129.36.92;tag=as6818347f
Contact: sip:asterisk@102.129.36.92:5060
Call-ID: 668b52770cfd48513553521920c04189@102.129.36.92:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.38.3
Date: Sat, 10 Sep 2022 16:14:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:194.28.167.186:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK6536f1dd;rport=5060;received=102.129.36.92
From: “asterisk” sip:asterisk@102.129.36.92;tag=as4a252aff
To: sip:194.28.167.186;tag=850384556
Call-ID: 1933ffc5788bff4b32a949371708afed@102.129.36.92:5060
CSeq: 102 OPTIONS
Server: MediaCore/3.0.0
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘1933ffc5788bff4b32a949371708afed@102.129.36.92:5060’ Method: OPTIONS
Reliably Transmitting (NAT) to 134.119.204.23:5060:
OPTIONS sip:134.119.204.23 SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK65f2ae03;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@102.129.36.92;tag=as7516bd6f
Contact: sip:asterisk@102.129.36.92:5060
Call-ID: 4dba1fa5755165fc4036747472d3073a@102.129.36.92:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.38.3
Date: Sat, 10 Sep 2022 16:14:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:194.28.167.31:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK73c76c4a;rport=5060;received=102.129.36.92
From: “asterisk” sip:asterisk@102.129.36.92;tag=as5f685ee4
To: sip:194.28.167.31;tag=214623864
Call-ID: 0da1ce7002aa7b302f989a0d420e7161@102.129.36.92:5060
CSeq: 102 OPTIONS
Server: MediaCore/3.0.0
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘0da1ce7002aa7b302f989a0d420e7161@102.129.36.92:5060’ Method: OPTIONS
<— SIP read from UDP:194.28.167.32:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK7f229723;rport=5060;received=102.129.36.92
From: “asterisk” sip:asterisk@102.129.36.92;tag=as6818347f
To: sip:194.28.167.32;tag=1375889022
Call-ID: 668b52770cfd48513553521920c04189@102.129.36.92:5060
CSeq: 102 OPTIONS
Server: MediaCore/3.0.0
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘668b52770cfd48513553521920c04189@102.129.36.92:5060’ Method: OPTIONS
<— SIP read from UDP:134.119.204.23:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK65f2ae03;rport=5060;received=102.129.36.92
From: “asterisk” sip:asterisk@102.129.36.92;tag=as7516bd6f
To: sip:134.119.204.23;tag=1255372405
Call-ID: 4dba1fa5755165fc4036747472d3073a@102.129.36.92:5060
CSeq: 102 OPTIONS
Server: MediaCore/3.0.0
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO, REGISTER
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘4dba1fa5755165fc4036747472d3073a@102.129.36.92:5060’ Method: OPTIONS
<— SIP read from UDP:197.210.19.91:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK3ca1351f;rport=5060
Call-ID: 0ad7b42a0945df613657e58c6375f0b9@102.129.36.92:5060
From: sip:+447749931908@102.129.36.92;tag=as513b94b1
To: sip:+2347038284899@197.210.19.91;tag=15152977913819
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Content-Length: 271
Content-Type: application/sdp
v=0
o=- 12396772 12396772 IN IP4 197.210.19.108
s=SBC call
c=IN IP4 197.210.19.108
t=0 0
a=sendrecv
m=audio 56148 RTP/AVP 8 101
c=IN IP4 197.210.19.108
b=RR:3000
b=RS:1000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:40
<------------->
— (11 headers 14 lines) —
sip_route_dump: route/path hop: sip:197.210.19.91:5060;transport=udp;lr;Hpt=nw_36a_631cc578_1743e11_ex_90f8_216;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=20984
Got SDP version 12396772 and unique parts [- 12396772 IN IP4 197.210.19.108]
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|alaw|ulaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
0x7ff56400ba50 – Strict RTP learning after remote address set to: 197.210.19.108:56148
Peer audio RTP is at port 197.210.19.108:56148
– SIP/mtn-00000005 is making progress passing it to SIP/5337260406-00000004
Audio is at 14984
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 197.211.58.14:37664 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—eeeb5c745b03db34;received=197.211.58.14;rport=37664
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
To: sip:2347038284899@102.129.36.92:5060;tag=as7fb89a5c
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 2 INVITE
Server: Asterisk PBX 13.38.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2347038284899@102.129.36.92:5060
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 1359784157 1359784157 IN IP4 102.129.36.92
s=Asterisk PBX 13.38.3
c=IN IP4 102.129.36.92
t=0 0
m=audio 14984 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
<------------>
<— SIP read from UDP:197.210.19.91:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK3ca1351f;rport=5060
Call-ID: 0ad7b42a0945df613657e58c6375f0b9@102.129.36.92:5060
From: sip:+447749931908@102.129.36.92;tag=as513b94b1
To: sip:+2347038284899@197.210.19.91;tag=15152977913819
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:197.210.19.91:5060;transport=udp;lr;Hpt=nw_36a_631cc578_1743e11_ex_90f8_216;CxtId=3;TRC=ffffffff-ffffffff;X-HwB2bUaCookie=20984
– SIP/mtn-00000005 is ringing
0x7ff56400ba50 – Strict RTP switching to RTP target address 197.210.19.108:56148 as source
0x7ff4e800bf50 – Strict RTP qualifying stream type: audio
0x7ff4e800bf50 – Strict RTP switching source address to 197.211.58.14:36703
0x7ff4e800bf50 – Strict RTP learning complete - Locking on source address 197.211.58.14:36703
<— SIP read from UDP:197.211.58.14:37664 —>
<------------->
0x7ff56400ba50 – Strict RTP learning complete - Locking on source address 197.210.19.108:56148
Really destroying SIP dialog ‘0ChtCRH2q9zMUi4zg1ET4Q…’ Method: REGISTER
<— SIP read from UDP:197.210.19.91:5060 —>
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK3ca1351f;rport=5060
Call-ID: 0ad7b42a0945df613657e58c6375f0b9@102.129.36.92:5060
From: sip:+447749931908@102.129.36.92;tag=as513b94b1
To: sip:+2347038284899@197.210.19.91;tag=15152977913819
CSeq: 102 INVITE
Reason: Q.850;cause=19
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 197.210.19.91:5060:
ACK sip:197.210.19.91:5060;transport=udp;Hpt=nw_36a_631cc578_1743e11_ex_90f8_16;CxtId=3;TRC=ffffffff-ffffffff SIP/2.0
Via: SIP/2.0/UDP 102.129.36.92:5060;branch=z9hG4bK3ca1351f;rport
Max-Forwards: 70
From: sip:+447749931908@102.129.36.92;tag=as513b94b1
To: sip:+2347038284899@197.210.19.91;tag=15152977913819
Contact: sip:+447749931908@102.129.36.92:5060
Call-ID: 0ad7b42a0945df613657e58c6375f0b9@102.129.36.92:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.38.3
Content-Length: 0
– SIP/mtn-00000005 redirecting info has changed, passing it to SIP/5337260406-00000004
<— Transmitting (NAT) to 197.211.58.14:37664 —>
SIP/2.0 181 Call is being forwarded
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—eeeb5c745b03db34;received=197.211.58.14;rport=37664
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
To: sip:2347038284899@102.129.36.92:5060;tag=as7fb89a5c
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 2 INVITE
Server: Asterisk PBX 13.38.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:2347038284899@102.129.36.92:5060
Content-Length: 0
<------------>
– SIP/mtn-00000005 is busy
Scheduling destruction of SIP dialog ‘0ad7b42a0945df613657e58c6375f0b9@102.129.36.92:5060’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:1/0/0)
– AGI Script Executing Application: (Busy) Options: (1)
<— Reliably Transmitting (NAT) to 197.211.58.14:37664 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—eeeb5c745b03db34;received=197.211.58.14;rport=37664
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
To: sip:2347038284899@102.129.36.92:5060;tag=as7fb89a5c
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 2 INVITE
Server: Asterisk PBX 13.38.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: User alerting, no answer
X-Asterisk-HangupCauseCode: 19
Content-Length: 0
<------------>
– <SIP/5337260406-00000004>AGI Script a2billing.php completed, returning 4
== Spawn extension (a2billing, 2347038284899, 18) exited non-zero on ‘SIP/5337260406-00000004’
<— SIP read from UDP:197.211.58.14:37664 —>
ACK sip:2347038284899@102.129.36.92:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:49749;branch=z9hG4bK-524287-1—eeeb5c745b03db34;rport
Max-Forwards: 70
To: sip:2347038284899@102.129.36.92:5060;tag=as7fb89a5c
From: sip:5337260406@102.129.36.92:5060;transport=UDP;tag=29655115
Call-ID: 24mhrqKqSTdrLzGpONTftg…
CSeq: 2 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘24mhrqKqSTdrLzGpONTftg…’ Method: ACK
Your provider is giving you a 480 just like you want. Callee didn’t answer so they gave a 480.
Asterisk is sending back a 181 Call is being forwarded to the caller based on getting a 480 from the Dial()'d channel.
Now Asterisk is telling your caller that everything is busy. This is probably the result of the 181 reply and that not being handled. Should that happen? Should there be a 181 reply at this point?
Your provider is giving you the right response, Asterisk is then doing something else based on that reply that causes your caller to get a 486 instead. Not sure what that could be since this is all based on things (Asterisk version, chan_sip, A2Billing) that are basically all EOL and really can’t be supported. Whatever is happening is happening based on the logic being used in Asterisk so far.
where could this 181 call forwarding be set up please?
can disabling NAT help?
I can’t tell you where this is setup. It’s your system and it has A2Billing on it, you’re going to need to look at this account/user and see if they have something setup they shouldn’t or what else is going on. If this is the only user having the problem, then it must be related to the settings for this user.
it affects all other user. I have two different trunks from 2 different providers and this happens when I call through those trunks with different users. I am getting frustrated. I need this fixed asap in anyway.
Does this only happen when they return a 480 or does it happen on other call statuses?
it happens only when they return 480
I’m thinking, could this be SIP ALG issue?
Asterisk is also indicating NO ANSWER, as the hangup cause, which should have been translated to 480.
However the 181 is totally weird. I can’t see that being sent unless the AGI script is doing something weird, My guess is that you’ve got some sort of mixed up status as a result of a failed attempt redirect the call within the script.
I also note that the dialplan Hangup isn’t being executed, which means that the AGI script has already hung up the channel.
At the moment, I would say this should be treated as an a2billing problem not an Asterisk one.
(Does a2billing possibly use a customised Asterisk?)
Nope. I am going with A2Billing issue as well, as Asterisk is what is generating the 181 on receiving a 480.
a2billing.php appears to be a very large script, spread over 11 files, and which hasn’t been touched in over a decade. I doubt that anyone is going to be able to sensibly address this in a reasonable amount of time unless they are already familiar with its internals.
talking of a2billing.php, I have another link that returns SIP 480, same a2billing and asterisk version is running on both. only difference is that this is connected to provider using E1 (dahdi). now, I want to copy it’s working a2billing.php and see if it works.