SIP/2.0 486 and 404 not found when trying outgoing call from asterisk

romal.amarkhail · April 3, 2019, 5:03pm

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 486 Busy Here
To: sip:7857272051@10.202.79.99:5060;tag=6f4381d949237404i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as3638137f
Call-ID: 4d5b0be53d60546a167394712451bfc2@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK61c36166
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

ambiorixg12 · April 3, 2019, 5:32pm

The trace is incomplete, but the SPA2102 is unable to take more calls


486 the callee's end system was contacted successfully, but the callee is
   currently not willing or able to take additional calls at this end
   system.

romal.amarkhail · April 3, 2019, 5:38pm

Thanks! I can receive calls but cant make any calls. Is there any solutions ?

romal.amarkhail · April 3, 2019, 5:39pm

The user end device is analog connected to an ATA, we replaced the ATA and tweak all the possible configs in ATA but still cant make any inbound or outbound calls.

ambiorixg12 · April 3, 2019, 5:44pm

ATA it is a FXS adapter so you just can use it in order to connect analog phones to Asterisk as SIP clients, post a full SIP trace for outbound when using the ATA

romal.amarkhail · April 3, 2019, 5:53pm

Can you please show me how to trace the outbound when using the ATA ? Do you mean to tracerout from Asterisk to ATA ?

ambiorixg12 · April 3, 2019, 6:17pm

sip set debug on or sip set debug peer name

romal.amarkhail · April 3, 2019, 6:51pm

Incoming call: Good

Reliably Transmitting (no NAT) to 64.189.129.133:24815:
OPTIONS sip:7857272051@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK351ebd11
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as4796492c
To: sip:7857272051@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 7f10da89760aefb9582124187c61b450@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:41:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 404 Not Found
To: sip:7857272051@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as4796492c
Call-ID: 7f10da89760aefb9582124187c61b450@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK351ebd11
Server: Linksys/SPA2102-3.3.6
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘7f10da89760aefb9582124187c61b450@64.189.248.66:5060’ Method: OPTIONS
[Apr 3 11:41:38] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:Ports@64.189.248.66;tag=1261249251
[Apr 3 11:41:38] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:iad733@64.189.248.66;tag=669214385
[Apr 3 11:41:38] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 1637686302-645863443-148670061 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 3 11:41:39] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:btc@64.189.248.66;tag=2103078328
[Apr 3 11:41:41] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 1527572019-2084484645-1421027402 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 3 11:41:45] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 819779398-2050699096-410919234 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 3 11:41:46] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 2001786299-1879367839-155724309 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 3 11:41:46] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 1685782249-755342636-1418283641 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
Reliably Transmitting (NAT) to 64.189.129.133:24815:
OPTIONS sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK0da606b0;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as4224eb3c
To: sip:7857272054@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 68acaefd3b8cfe40148b097b722587a2@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:41:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 200 OK
To: sip:7857272054@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as4224eb3c
Call-ID: 68acaefd3b8cfe40148b097b722587a2@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK0da606b0
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘68acaefd3b8cfe40148b097b722587a2@64.189.248.66:5060’ Method: OPTIONS
[Apr 3 11:41:50] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:RDEVICE7@64.189.248.66;tag=535258257
RI-server2*CLI> exit
Executing last minute cleanups
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]#
[root@RI-server2 ~]# asterisk -rvvvvvvvvvvvvv
Asterisk 1.8.9.3, Copyright © 1999 - 2011 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

== Parsing ‘/etc/asterisk/asterisk.conf’: == Found
== Parsing ‘/etc/asterisk/extconfig.conf’: == Found
Connected to Asterisk 1.8.9.3 currently running on RI-server2 (pid = 21921)
Verbosity was 10 and is now 13
RI-server2CLI>
RI-server2CLI>
RI-server2CLI>
RI-server2CLI>
RI-server2CLI>
RI-server2CLI>
[Apr 3 11:47:28] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:zPBX4@64.189.248.66;tag=411848048
[Apr 3 11:47:28] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 154113914-520342387-332648914 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Apr 3 11:47:28] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:DataCAD@64.189.248.66;tag=1045796729
[Apr 3 11:47:34] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:lineS2@64.189.248.66;tag=500087475
[Apr 3 11:47:36] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:incredibly@64.189.248.66;tag=799603174
== Using SIP RTP CoS mark 5
– Executing [+17857272054@from-pstn:1] Macro(“SIP/bandwidth-1-00000171”, “stdexten,7857272054,sip/7857272054”) in new stack
– Executing [s@macro-stdexten:1] Set(“SIP/bandwidth-1-00000171”, “dynext=”) in new stack
– Executing [s@macro-stdexten:2] NoOp(“SIP/bandwidth-1-00000171”, “”) in new stack
– Executing [s@macro-stdexten:3] NoOp(“SIP/bandwidth-1-00000171”, “0”) in new stack
– Executing [s@macro-stdexten:4] GotoIf(“SIP/bandwidth-1-00000171”, “0?s,100”) in new stack
– Executing [s@macro-stdexten:5] GotoIf(“SIP/bandwidth-1-00000171”, “0?s,100”) in new stack
– Executing [s@macro-stdexten:6] GotoIf(“SIP/bandwidth-1-00000171”, “0?s,100”) in new stack
– Executing [s@macro-stdexten:7] GotoIf(“SIP/bandwidth-1-00000171”, “0?s,200”) in new stack
– Executing [s@macro-stdexten:8] GotoIf(“SIP/bandwidth-1-00000171”, “1?s,300”) in new stack
– Goto (macro-stdexten,s,300)
– Executing [s@macro-stdexten:300] Set(“SIP/bandwidth-1-00000171”, “dynext=7857272054”) in new stack
– Executing [s@macro-stdexten:301] Goto(“SIP/bandwidth-1-00000171”, “dial”) in new stack
– Goto (macro-stdexten,s,9)
– Executing [s@macro-stdexten:9] Dial(“SIP/bandwidth-1-00000171”, “SIP/7857272054,20”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19984
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.189.129.133:24815:
INVITE sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK728bbb88;rport
Max-Forwards: 70
From: “WIRELESS CALLER” sip:+17372032268@64.189.248.66;tag=as02db7972
To: sip:7857272054@64.189.129.133:24815
Contact: sip:+17372032268@64.189.248.66:5060
Call-ID: 3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:47:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 476

v=0
o=root 1494499645 1494499645 IN IP4 64.189.248.66
s=Asterisk PBX 1.8.9.3
c=IN IP4 64.189.248.66
t=0 0
m=audio 19984 RTP/AVP 0 3 8 112 5 10 7 111 9 118 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called SIP/7857272054

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 100 Trying
To: sip:7857272054@64.189.129.133:24815
From: “WIRELESS CALLER” sip:+17372032268@64.189.248.66;tag=as02db7972
Call-ID: 3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK728bbb88
Server: Linksys/SPA2102-3.3.6
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 180 Ringing
To: sip:7857272054@64.189.129.133:24815;tag=8cd8b3669e2ba5e9i0
From: “WIRELESS CALLER” sip:+17372032268@64.189.248.66;tag=as02db7972
Call-ID: 3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK728bbb88
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 7857272054 sip:7857272054@64.189.248.66;screen=yes;party=called
Content-Length: 0

<------------->
— (9 headers 0 lines) —
list_route: no route
– SIP/7857272054-00000172 is ringing
Reliably Transmitting (no NAT) to 64.189.129.133:24815:
OPTIONS sip:7857272051@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK74a966b3
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as0ce80b89
To: sip:7857272051@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 27fabd4676b9be0e67dbe82e707aac2a@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:47:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 404 Not Found
To: sip:7857272051@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as0ce80b89
Call-ID: 27fabd4676b9be0e67dbe82e707aac2a@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK74a966b3
Server: Linksys/SPA2102-3.3.6
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘27fabd4676b9be0e67dbe82e707aac2a@64.189.248.66:5060’ Method: OPTIONS
[Apr 3 11:47:38] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 572860650-1565386819-2013153263 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
[Apr 3 11:47:39] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device 1001sip:1001@64.189.248.66;tag=0a38d710

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 200 OK
To: sip:7857272054@64.189.129.133:24815;tag=8cd8b3669e2ba5e9i0
From: “WIRELESS CALLER” sip:+17372032268@64.189.248.66;tag=as02db7972
Call-ID: 3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK728bbb88
Contact: 7857272054 sip:7857272054@64.189.129.133:24815
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 7857272054 sip:7857272054@64.189.248.66;screen=yes;party=called
Content-Length: 255
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 65432 65432 IN IP4 64.189.129.133
s=-
c=IN IP4 64.189.129.133
t=0 0
m=audio 23928 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
— (13 headers 13 lines) —
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.189.129.133:23928
list_route: hop: sip:7857272054@64.189.129.133:24815
set_destination: Parsing sip:7857272054@64.189.129.133:24815 for address/port to send to
set_destination: set destination to 64.189.129.133:24815
Transmitting (NAT) to 64.189.129.133:24815:
ACK sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK7603c486;rport
Max-Forwards: 70
From: “WIRELESS CALLER” sip:+17372032268@64.189.248.66;tag=as02db7972
To: sip:7857272054@64.189.129.133:24815;tag=8cd8b3669e2ba5e9i0
Contact: sip:+17372032268@64.189.248.66:5060
Call-ID: 3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.9.3
Content-Length: 0

-- SIP/7857272054-00000172 answered SIP/bandwidth-1-00000171
-- Locally bridging SIP/bandwidth-1-00000171 and SIP/7857272054-00000172

[Apr 3 11:47:41] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:incredibly@64.189.248.66;tag=529139654
[Apr 3 11:47:45] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 1866289166-1538387655-968676845 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 3 11:47:46] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:lineS2@64.189.248.66;tag=1453044951
[Apr 3 11:47:46] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:Synsip@64.189.248.66;tag=181138959
Reliably Transmitting (NAT) to 64.189.129.133:24815:
OPTIONS sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK1dd4d215;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as052b5b71
To: sip:7857272054@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 2197568c25eb82730be518607028946b@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:47:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 200 OK
To: sip:7857272054@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as052b5b71
Call-ID: 2197568c25eb82730be518607028946b@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK1dd4d215
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘2197568c25eb82730be518607028946b@64.189.248.66:5060’ Method: OPTIONS
[Apr 3 11:47:48] ERROR[22873]: cdr_mysql.c:342 mysql_log: Failed to insert into database: (1062) Duplicate entry ‘’ for key ‘accountcode’
Scheduling destruction of SIP dialog ‘3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:7857272054@64.189.129.133:24815 for address/port to send to
set_destination: set destination to 64.189.129.133:24815
Reliably Transmitting (NAT) to 64.189.129.133:24815:
BYE sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK1ee6c5ec;rport
Max-Forwards: 70
From: “WIRELESS CALLER” sip:+17372032268@64.189.248.66;tag=as02db7972
To: sip:7857272054@64.189.129.133:24815;tag=8cd8b3669e2ba5e9i0
Call-ID: 3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.9.3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (macro-stdexten, s, 9) exited non-zero on ‘SIP/bandwidth-1-00000171’ in macro ‘stdexten’
== Spawn extension (from-pstn, +17857272054, 1) exited non-zero on ‘SIP/bandwidth-1-00000171’

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 200 OK
To: sip:7857272054@64.189.129.133:24815;tag=8cd8b3669e2ba5e9i0
From: “WIRELESS CALLER” sip:+17372032268@64.189.248.66;tag=as02db7972
Call-ID: 3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060
CSeq: 103 BYE
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK1ee6c5ec
Server: Linksys/SPA2102-3.3.6
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘3645b4f75b01d04d57d6b50d21605c22@64.189.248.66:5060’ Method: INVITE
RI-server2*CLI> exit
Executing last minute cleanups

=============================================================================================================

Outgoing call: Not good

Reliably Transmitting (no NAT) to 64.189.129.133:24815:
OPTIONS sip:7857272051@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK78adfcea
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as545e2f7b
To: sip:7857272051@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 300409316e4d569754e28daf30686465@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:48:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 404 Not Found
To: sip:7857272051@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as545e2f7b
Call-ID: 300409316e4d569754e28daf30686465@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK78adfcea
Server: Linksys/SPA2102-3.3.6
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘300409316e4d569754e28daf30686465@64.189.248.66:5060’ Method: OPTIONS
[Apr 3 11:48:38] NOTICE[21935]: chan_sip.c:14259 check_auth: Correct auth, but based on stale nonce received from ‘sip:4047935672@64.189.248.66;tag=a9d2cd8c8e5fee07o0’
[Apr 3 11:48:38] NOTICE[21935]: chan_sip.c:20605 handle_response_peerpoke: Peer ‘4047935672’ is now Reachable. (103ms / 2000ms)
[Apr 3 11:48:43] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:Container@64.189.248.66;tag=135609233
Reliably Transmitting (NAT) to 64.189.129.133:24815:
OPTIONS sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK31c03353;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as7c6fe1b5
To: sip:7857272054@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 07a2eb161656b5d6753fd7083d443351@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:48:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 486 Busy Here
To: sip:7857272054@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as7c6fe1b5
Call-ID: 07a2eb161656b5d6753fd7083d443351@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK31c03353
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘07a2eb161656b5d6753fd7083d443351@64.189.248.66:5060’ Method: OPTIONS
[Apr 3 11:48:47] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:Container@64.189.248.66;tag=991393627
[Apr 3 11:48:48] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:DISCOUNTS@64.189.248.66;tag=317244230
[Apr 3 11:48:48] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 225631811-725946833-1023681060 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 3 11:48:49] WARNING[21935]: chan_sip.c:3625 retrans_pkt: Retransmission timeout reached on transmission 1469374468-1007958627-1522894852 for seqno 1 (Non-critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Apr 3 11:48:51] NOTICE[21935]: chan_sip.c:22358 handle_request_invite: Sending fake auth rejection for device sip:Udevice9@64.189.248.66;tag=2044259123
RI-server2*CLI> exit
Executing last minute cleanups
[root@RI-server2 ~]#

ambiorixg12 · April 3, 2019, 6:59pm

The 486 reply, it is in response to an OPTION request, make an outbound call and post the whole trace starting fro the INVITE request

romal.amarkhail:

OPTIONS sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK31c03353;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as7c6fe1b5
To: sip:7857272054@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 07a2eb161656b5d6753fd7083d443351@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 18:48:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 486 Busy Here
To: sip:7857272054@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as7c6fe1b5
Call-ID: 07a2eb161656b5d6753fd7083d443351@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK31c03353
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

romal.amarkhail · April 3, 2019, 7:37pm

Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK5a6d6a44;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as6c1b86d4
To: sip:7857272054@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 424b5778620b680a00912ce24a59634a@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 19:29:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 486 Busy Here
To: sip:7857272054@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as6c1b86d4
Call-ID: 424b5778620b680a00912ce24a59634a@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK5a6d6a44
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

<------------->
— (10 headers 0 lines) —

Reliably Transmitting (NAT) to 64.189.129.133:24815:
OPTIONS sip:7857272054@64.189.129.133:24815 SIP/2.0
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK0710cc49;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@64.189.248.66;tag=as3822e579
To: sip:7857272054@64.189.129.133:24815
Contact: sip:asterisk@64.189.248.66:5060
Call-ID: 05e7816c0971d8ac72a264a95c49f349@64.189.248.66:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.9.3
Date: Wed, 03 Apr 2019 19:30:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:64.189.129.133:24815 —>
SIP/2.0 200 OK
To: sip:7857272054@64.189.129.133:24815;tag=b35e9b411b50e9d0i0
From: “asterisk” sip:asterisk@64.189.248.66;tag=as3822e579
Call-ID: 05e7816c0971d8ac72a264a95c49f349@64.189.248.66:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.189.248.66:5060;branch=z9hG4bK0710cc49
Server: Linksys/SPA2102-3.3.6
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura

romal.amarkhail · April 3, 2019, 8:13pm

Btw this is an analog phone connected to a Cisco ATA adapter. Could be something working with my extension.conf ?

[general]

static=yes ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
DYNAMIC_FEATURES=>automon

;=============================================
; Apogee Johnson and Wales University
;
;=============================================

[default]

include => from-sip

[bogon-calls]

exten => _x.,1,Congestion

;===============================================================

;==============================================
; collage village safety line
;===============================================

[college-village-safety]

exten => _15858139262,1,noop(–cid_number–{CALLERID(num)}-------cid_name--{C$
exten => _15858139262,n,dial(sip/bandwidth-1/+{EXTEN}) exten => _15858139262,n,dial(sip/bandwidth-2/+{EXTEN})
exten => _15858139262,n,congestion

exten => _5858139262,1,noop(–cid_number–{CALLERID(num)}-------cid_name--{C$
exten => _5858139262,n,dial(sip/bandwidth-1/+1${EXTEN})
exten => _5858139262,n,dial(sip/bandwidth-2/+1${EXTEN})
exten => _5858139262,n,congestion

include => 911-college-village-safety

;================================================

[macro-stdexten]

exten => s,1,Set(dynext={DB(dynext/{ARG1})})
exten => s,n,NoOp({dynext}) exten => s,n,NoOp({LEN({dynext})}) exten => s,n,GotoIf(["{LEN({dynext})}" = “7”]?s,100)
exten => s,n,GotoIf(["{LEN({dynext})}" = "10"]?s,100) exten => s,n,GotoIf(["{LEN({dynext})}" = “11”]?s,100)
exten => s,n,GotoIf(["{LEN({dynext})}" = "6"]?s,200) ; Calls 6-digit Extension exten => s,n,GotoIf(["{LEN({dynext})}" = “0”]?s,300) ; Unspecified
exten => s,n(dial),Dial(SIP/{dynext},20) ; Ring the interface, 20 seconds maximum exten => s,n,Goto(s-{DIALSTATUS},1) ; Jump based on status
exten => s,102,Goto(s-{DIALSTATUS},1) ; Jump based on status exten => s,200,Goto(from-sip,{dynext},1); Calls 6-digit Extension
exten => s,300,Set(dynext=${ARG1})
exten => s,301,Goto(dial)

exten => s-BUSY,1,Voicemail({ARG1},b) ; If busy, send to voicemail w/ busy announce exten => s-BUSY,2,Hangup exten => s-NOANSWER,1,Voicemail({ARG1},u) ; If unavailable, send to voicemail
exten => s-NOANSWER,2,Hangup
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1})
exten => a,2,hangup

;===================================================================

[911-college-village-safety]

exten => 911,1,set(CALLERID(num)=5858616880)
exten => 911,n,set(CALLERID(ani)=5858616880)
exten => 911,n,dial(sip/bandwidth-1/911)
exten => 911,n,dial(sip/bandwidth-2/911)
exten => 911,n,Hangup()

exten => 9911,1,goto(911-college-village-safety,911,1)

[911]

exten => 911,1,dial(sip/bandwidth-1/911)
exten => 911,n,dial(sip/bandwidth-2/911)
exten => 911,n,Hangup()

exten => 9911,1,goto(911,911,1)

[from-sip]

include => 911
include => internal
include => outbound
include => utilities

[internal]

exten => _+1NXXNXXXXXX,1,Macro(stdexten,{EXTEN:2},sip/{EXTEN:2})

;exten => _+17857272054,1,answer()
;exten => _+17857272054,n,disa(no-password,from-sip)

[test-context]
exten => s,1,answer
exten => s,n,set(CALLERID(num)=4105616070)
exten => s,n,set(CALLERID(ani)=4105616070)
exten => s,n,goto(911,911,1)

[from-pstn]

; safety line
;=======================
exten => +15858616880,1,NoOp(==== Dialing Safety Line ====)
exten => +15858616880,n,Macro(stdexten,0163,0163)

;=======================

;test number
exten => _+17857272060,1,answer
exten => _+17857272060,n,authenticate(5678)
exten => _+17857272060,n,set(CALLERID(num)=)
exten => _+17857272060,n,set(CALLERID(ani)=)
exten => _+17857272060,n,goto(911-college-village-safety,911,1)

include => internal

[outbound]

; LOCAL - outbound

exten => _NXXXXXX,1,goto(outbound,+1785${EXTEN},1)

; 800 TOLL FREE - outbound

exten => _8XXNXXXXXX,1,goto(outbound,1${EXTEN},1)

exten => _18XXNXXXXXX,1,noop(–cid_number–{CALLERID(num)}-------cid_name--{CALLERID(name)}–)
exten => _18XXNXXXXXX,n,Dial(sip/bandwidth-1/+{EXTEN},,) exten => _18XXNXXXXXX,n,Dial(sip/bandwidth-2/+{EXTEN},)
exten => _18XXNXXXXXX,n,Congestion

; LONG DISTANCE - outbound

exten => _NXXNXXXXXX,1,goto(outbound,1${EXTEN},1)

exten => _1NXXNXXXXXX,1,noop(–cid_number–{CALLERID(num)}-------cid_name--{CALLERID(name)}–)
exten => _1NXXNXXXXXX,n,dial(sip/bandwidth-1/+{EXTEN}) exten => _1NXXNXXXXXX,n,dial(sip/bandwidth-2/+{EXTEN})
exten => _1NXXNXXXXXX,n,congestion

; INTERNATIONAL - outbound

;exten => _011.,1,Dial(sip/bandwidth-1/{EXTEN}) ;exten => _011.,n,Dial(sip/bandwidth-2/{EXTEN})
;exten => _011.,n,Congestion

;exten => _022.,1,Dial(sip/bandwidth-1/{EXTEN}) ;exten => _022.,n,Dial(sip/bandwidth-2/{EXTEN})
;exten => _022.,n,Congestion

; 411 - outbound

exten => 411,1,Dial(sip/bandwidth-1/{EXTEN}) exten => 411,n,Dial(sip/bandwidth-2/{EXTEN})
exten => 411,n,Congestion

;==================================================

[utilities]

;Voicemail access
exten => *98,1,answer
exten => *98,n,wait(1)
exten => *98,n,voicemailmain(${CALLERID(num))
exten => *98,n,hangup

;Music-on-Hold test

exten => 811,1,WaitMusicOnHold(60)
exten => 811,2,Hangup

;MeetMe Conferances

exten => 0100,1,answer
exten => 0100,2,wait(2)
exten => 0100,3,authenticate(1097)
exten => 0100,4,MeetMe(0100||)
exten => 0100,5,Hangup

exten => 0101,1,answer
exten => 0101,2,wait(2)
exten => 0101,3,authenticate(1097)
exten => 0101,4,MeetMe(0101||)
exten => 0101,5,Hangup

exten => 0102,1,answer
exten => 0102,2,wait(2)
exten => 0102,3,authenticate(1097)
exten => 0102,4,MeetMe(0102||)
exten => 0102,5,Hangup

exten => 0103,1,answer
exten => 0103,2,wait(2)
exten => 0103,3,authenticate(1097)
exten => 0103,4,MeetMe(0103||)
exten => 0103,5,Hangup

;
;Call Park
;
include => parkedcalls

;
; Record/set Holiday Closed Message
;
exten => 897,1,authenticate(5678)
exten => 897,n,Wait(2)
exten => 897,n,Record(custom/closed_message:gsm)
exten => 897,n,Wait(2)
exten => 897,n,DBput(closed/1=1)
exten => 897,n,Playback(custom/closed_message)
exten => 897,n,wait(2)
exten => 897,n,Hangup

;
; Disable holiday greeting
;
exten => 898,1,authenticate(5678)
exten => 898,n,Wait(2)
exten => 898,n,DBput(closed/1=0)
exten => 898,n,Playback(beep)
exten => 898,n,wait(2)
exten => 898,n,Hangup

;
; Record voice file to /tmp directory
;
exten => 899,1,answer()
exten => 899,n,Wait(2)
exten => 899,n,Record(audio-test:gsm)
exten => 899,n,Wait(2)
exten => 899,n,Playback(audio-test)
exten => 899,n,wait(2)
exten => 899,n,Hangup

exten => 832,1,authenticate(5678)
exten => 832,2,ChanSpy(agent)
exten => 832,3,Hangup