One way Audio with odbc users

wesly1988 · April 12, 2019, 1:54am

Hi Guy,

Got a strange problem.
Trying to call from a sip client to a normal phone or exetension.
This results always in a one way audio connenction.

I use the odbc database, and can’t really find the problem.
Can anybody help me in the right direction.
There seems to be no errors at all.

[general]
context=public
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0:15060
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
language=ja
externaddr=52.194.253.25
localnet=172.31.29.32/255.255.240.0
nat=force_rport,comedia
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm

/var/log/asterisk/messages

[Apr 12 10:44:36] VERBOSE[23055][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824@context_tok:1] NoOp("SIP/inbound_1_1-00000003", "inbound") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824@context_tok:2] Dial("SIP/inbound_1_1-00000003", "SIP/1_1_1_1/1_1_1_1&SIP/1_1_1_2/1_1_1_2&SIP/1_1_1_3/1_1_1_3&SIP/1_1_1_4/1_1_1_4") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_1/1_1_1_1
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_3/1_1_1_3
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_1-00000004 is ringing
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 is ringing
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 answered SIP/inbound_1_1-00000003
[Apr 12 10:44:44] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] pbx.c: Spawn extension (context_tok, 52431824, 2) exited non-zero on 'SIP/inbound_1_1-00000003'

<------------>
    -- Executing [52431824@context_tok:1] NoOp("SIP/inbound_1_1-00000003", "inbound") in new stack
    -- Executing [52431824@context_tok:2] Dial("SIP/inbound_1_1-00000003", "SIP/1_1_1_1/1_1_1_1&SIP/1_1_1_2/1_1_1_2&SIP/1_1_1_3/1_1_1_3&SIP/1_1_1_4/1_1_1_4") in new stack
  == Using SIP RTP CoS mark 5
[Apr 12 11:00:44] WARNING[2405][C-00000001]: app_dial.c:2432 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  == Using SIP RTP CoS mark 5
[Apr 12 11:00:44] WARNING[2405][C-00000001]: app_dial.c:2432 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Audio is at 20688
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 111.108.30.208:60463:
INVITE sip:1_1_1_1@111.108.30.208:60463 SIP/2.0
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK5b8a3ae1;rport
Max-Forwards: 70
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as0dbab8bf
To: <sip:1_1_1_1@111.108.30.208:60463>
Contact: <sip:05052424142@52.194.253.25:15060>
Call-ID: 0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.11.0-rc1
Date: Fri, 12 Apr 2019 02:00:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 1567942082 1567942082 IN IP4 52.194.253.25
s=Asterisk PBX 13.11.0-rc1
c=IN IP4 52.194.253.25
t=0 0
m=audio 20688 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=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/1_1_1_1/1_1_1_1
Audio is at 20836
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 111.108.30.208:60645:
INVITE sip:1_1_1_3@111.108.30.208:60645 SIP/2.0
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK47c464e7;rport
Max-Forwards: 70
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as3c6244e4
To: <sip:1_1_1_3@111.108.30.208:60645>
Contact: <sip:05052424142@52.194.253.25:15060>
Call-ID: 467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.11.0-rc1
Date: Fri, 12 Apr 2019 02:00:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 1060488612 1060488612 IN IP4 52.194.253.25
s=Asterisk PBX 13.11.0-rc1
c=IN IP4 52.194.253.25
t=0 0
m=audio 20836 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=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/1_1_1_3/1_1_1_3
Really destroying SIP dialog '37f69b737b42ac1f4ac663ee4d8f6591@172.31.29.32:15060' Method: INVITE
Really destroying SIP dialog '6a54b1b155a136004aeedfb75cd01a39@172.31.29.32:15060' Method: INVITE

<--- SIP read from UDP:111.108.30.208:60645 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK47c464e7;rport=15060
To: <sip:1_1_1_3@111.108.30.208:60645>
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as3c6244e4
Call-ID: 467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:111.108.30.208:60463 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK5b8a3ae1;rport=15060
To: <sip:1_1_1_1@111.108.30.208:60463>
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as0dbab8bf
Call-ID: 0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:111.108.30.208:60463 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK5b8a3ae1;rport=15060
Contact: <sip:1_1_1_1@111.108.30.208:60463;transport=UDP>
To: <sip:1_1_1_1@111.108.30.208:60463>;tag=d775081a
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as0dbab8bf
Call-ID: 0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.2.25 rv2.8.112
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1_1_1_1@111.108.30.208:60463;transport=UDP>
    -- SIP/1_1_1_1-00000004 is ringing

<--- Transmitting (NAT) to 61.213.230.153:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 61.213.230.153:5060;branch=z9hG4bK09e98490;received=61.213.230.153;rport=5060
From: "05052424142" <sip:05052424142@smart.0038.net>;tag=as7d255076
To: <sip:52431824@52.194.253.25:15060>;tag=as303506f7
Call-ID: 5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp
CSeq: 102 INVITE
Server: Asterisk PBX 13.11.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:52431824@52.194.253.25:15060>
Content-Length: 0


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

<--- SIP read from UDP:61.213.230.153:5060 --->
INVITE sip:52431824@52.194.253.25:15060 SIP/2.0
Via: SIP/2.0/UDP 61.213.230.153:5060;branch=z9hG4bK09e98490;rport
Max-Forwards: 70
From: "05052424142" <sip:05052424142@smart.0038.net>;tag=as7d255076
To: <sip:52431824@52.194.253.25:15060>
Contact: <sip:05052424142@61.213.230.153:5060>
Call-ID: 5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp
CSeq: 102 INVITE
User-Agent: Fusion Open MGW 1.0
Date: Fri, 12 Apr 2019 02:00:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 338

v=0
o=root 1642396121 1642396121 IN IP4 61.213.230.9
s=-
c=IN IP4 61.213.230.9
t=0 0
m=audio 16664 RTP/AVP 0 110 97 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 16 lines) ---
Ignoring this INVITE request

<--- Transmitting (NAT) to 61.213.230.153:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 61.213.230.153:5060;branch=z9hG4bK09e98490;received=61.213.230.153;rport=5060
From: "05052424142" <sip:05052424142@smart.0038.net>;tag=as7d255076
To: <sip:52431824@52.194.253.25:15060>
Call-ID: 5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp
CSeq: 102 INVITE
Server: Asterisk PBX 13.11.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:52431824@52.194.253.25:15060>
Content-Length: 0


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

<--- SIP read from UDP:111.108.30.208:60645 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK47c464e7;rport=15060
Contact: <sip:1_1_1_3@111.108.30.208:60645;transport=UDP>
To: <sip:1_1_1_3@111.108.30.208:60645>;tag=cc38480f
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as3c6244e4
Call-ID: 467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.2.28 rv2.8.115
Allow-Events: presence, kpml, talk
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:1_1_1_3@111.108.30.208:60645;transport=UDP>
    -- SIP/1_1_1_3-00000005 is ringing
Reliably Transmitting (NAT) to 61.213.230.153:5060:
OPTIONS sip:smart.0038.net SIP/2.0
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK7d93a120;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@52.194.253.25:15060>;tag=as6054f0b6
To: <sip:smart.0038.net>
Contact: <sip:asterisk@52.194.253.25:15060>
Call-ID: 11a86fb7797b113f4ed80efd7985cf9c@52.194.253.25:15060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.11.0-rc1
Date: Fri, 12 Apr 2019 02:00:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
       > 0x7f748401a0d0 -- Probation passed - setting RTP source address to 111.108.30.208:63507

<--- SIP read from UDP:111.108.30.208:60645 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK47c464e7;rport=15060
Contact: <sip:1_1_1_3@111.108.30.208:60645;transport=UDP>
To: <sip:1_1_1_3@111.108.30.208:60645>;tag=cc38480f
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as3c6244e4
Call-ID: 467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.2.28 rv2.8.115
Allow-Events: presence, kpml, talk
Content-Length: 606

v=0
o=Z 0 1 IN IP4 192.168.100.231
s=Z
c=IN IP4 192.168.100.231
t=0 0
m=audio 8000 RTP/AVP 0 106 9 3 111 8 97 110 112 102 101 98 100 99
a=rtpmap:106 opus/48000/2
a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
a=rtpmap:111 speex/16000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:112 speex/32000
a=rtpmap:102 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:100 telephone-event/16000
a=fmtp:100 0-16
a=rtpmap:99 telephone-event/32000
a=fmtp:99 0-16
a=sendrecv
<------------->
--- (12 headers 23 lines) ---
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 3
Found RTP audio format 111
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 112
Found RTP audio format 102
Found RTP audio format 101
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 99
Found audio description format opus for ID 106
Found audio description format speex for ID 111
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 112
Found audio description format G726-32 for ID 102
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Found unknown media description format telephone-event for ID 100
Found unknown media description format telephone-event for ID 99
Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw|gsm|alaw|g722|ilbc|g726|opus|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.100.231:8000
sip_route_dump: route/path hop: <sip:1_1_1_3@111.108.30.208:60645;transport=UDP>
Transmitting (NAT) to 111.108.30.208:60645:
ACK sip:1_1_1_3@111.108.30.208:60645;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK681344dd;rport
Max-Forwards: 70
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as3c6244e4
To: <sip:1_1_1_3@111.108.30.208:60645>;tag=cc38480f
Contact: <sip:05052424142@52.194.253.25:15060>
Call-ID: 467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.11.0-rc1
Content-Length: 0


---
    -- SIP/1_1_1_3-00000005 answered SIP/inbound_1_1-00000003
Scheduling destruction of SIP dialog '0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 111.108.30.208:60463:
CANCEL sip:1_1_1_1@111.108.30.208:60463 SIP/2.0
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK5b8a3ae1;rport
Max-Forwards: 70
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as0dbab8bf
To: <sip:1_1_1_1@111.108.30.208:60463>
Call-ID: 0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 13.11.0-rc1
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060' in 32000 ms (Method: INVITE)
Audio is at 20588
Adding codec ulaw to SDP

<--- Reliably Transmitting (NAT) to 61.213.230.153:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 61.213.230.153:5060;branch=z9hG4bK09e98490;received=61.213.230.153;rport=5060
From: "05052424142" <sip:05052424142@smart.0038.net>;tag=as7d255076
To: <sip:52431824@52.194.253.25:15060>;tag=as303506f7
Call-ID: 5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp
CSeq: 102 INVITE
Server: Asterisk PBX 13.11.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:52431824@52.194.253.25:15060>
Content-Type: application/sdp
Require: timer
Content-Length: 200

v=0
o=root 389784561 389784561 IN IP4 52.194.253.25
s=Asterisk PBX 13.11.0-rc1
c=IN IP4 52.194.253.25
t=0 0
m=audio 20588 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/1_1_1_3-00000005 joined 'simple_bridge' basic-bridge <b215c527-5737-4b21-a49e-8dc683225672>
    -- Channel SIP/inbound_1_1-00000003 joined 'simple_bridge' basic-bridge <b215c527-5737-4b21-a49e-8dc683225672>

<--- SIP read from UDP:111.108.30.208:60463 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK5b8a3ae1;rport=15060
Contact: <sip:1_1_1_1@111.108.30.208:60463;transport=UDP>
To: <sip:1_1_1_1@111.108.30.208:60463>;tag=d775081a
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as0dbab8bf
Call-ID: 0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060
CSeq: 102 CANCEL
User-Agent: Z 5.2.25 rv2.8.112
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:111.108.30.208:60463 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK5b8a3ae1;rport=15060
To: <sip:1_1_1_1@111.108.30.208:60463>;tag=d775081a
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as0dbab8bf
Call-ID: 0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060
CSeq: 102 INVITE
User-Agent: Z 5.2.25 rv2.8.112
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 111.108.30.208:60463:
ACK sip:1_1_1_1@111.108.30.208:60463;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK5b8a3ae1;rport
Max-Forwards: 70
From: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as0dbab8bf
To: <sip:1_1_1_1@111.108.30.208:60463>;tag=d775081a
Contact: <sip:05052424142@52.194.253.25:15060>
Call-ID: 0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.11.0-rc1
Content-Length: 0


---
Scheduling destruction of SIP dialog '0ecae5a91943906d1504bb3307d4e3ba@52.194.253.25:15060' in 32000 ms (Method: INVITE)
       > 0x7f748401a0d0 -- Probation passed - setting RTP source address to 111.108.30.208:63507
       > 0x7f748401a0d0 -- Probation passed - setting RTP source address to 111.108.30.208:63507

<--- SIP read from UDP:61.213.230.153:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK7d93a120;rport
From: "asterisk" <sip:asterisk@52.194.253.25:15060>;tag=as6054f0b6
To: <sip:smart.0038.net>;tag=as260fa312
Call-ID: 11a86fb7797b113f4ed80efd7985cf9c@52.194.253.25:15060
CSeq: 102 OPTIONS
Accept: application/sdp
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[Apr 12 11:00:46] NOTICE[1734]: chan_sip.c:24405 handle_response_peerpoke: Peer 'inbound_1_1' is now Reachable. (868ms / 2000ms)
Really destroying SIP dialog '11a86fb7797b113f4ed80efd7985cf9c@52.194.253.25:15060' Method: OPTIONS

<--- SIP read from UDP:61.213.230.153:5060 --->
ACK sip:52431824@52.194.253.25:15060 SIP/2.0
Via: SIP/2.0/UDP 61.213.230.153:5060;branch=z9hG4bK04d7e414;rport
Max-Forwards: 70
From: "05052424142" <sip:05052424142@smart.0038.net>;tag=as7d255076
To: <sip:52431824@52.194.253.25:15060>;tag=as303506f7
Contact: <sip:05052424142@61.213.230.153:5060>
Call-ID: 5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp
CSeq: 102 ACK
User-Agent: Fusion Open MGW 1.0
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
       > 0x7f747c0310f0 -- Probation passed - setting RTP source address to 61.213.230.9:16664
Really destroying SIP dialog '0552db525a3567f60b50687a6509e4b1@fkr13.fusioncom.co.jp' Method: BYE

<--- SIP read from UDP:111.108.30.208:60645 --->
BYE sip:05052424142@52.194.253.25:15060 SIP/2.0
Via: SIP/2.0/UDP 111.108.30.208:60645;branch=z9hG4bK-524287-1---acf29db1b52a666f;rport
Max-Forwards: 70
Contact: <sip:1_1_1_3@111.108.30.208:60645;transport=UDP>
To: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as3c6244e4
From: <sip:1_1_1_3@111.108.30.208:60645>;tag=cc38480f
Call-ID: 467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060
CSeq: 2 BYE
User-Agent: Z 5.2.28 rv2.8.115
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 111.108.30.208:60645 (NAT)
Scheduling destruction of SIP dialog '467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 111.108.30.208:60645 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.108.30.208:60645;branch=z9hG4bK-524287-1---acf29db1b52a666f;received=111.108.30.208;rport=60645
From: <sip:1_1_1_3@111.108.30.208:60645>;tag=cc38480f
To: "05052424142" <sip:05052424142@52.194.253.25:15060>;tag=as3c6244e4
Call-ID: 467fcf48607c230d7c495b3f6b540dc5@52.194.253.25:15060
CSeq: 2 BYE
Server: Asterisk PBX 13.11.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/1_1_1_3-00000005 left 'simple_bridge' basic-bridge <b215c527-5737-4b21-a49e-8dc683225672>
    -- Channel SIP/inbound_1_1-00000003 left 'simple_bridge' basic-bridge <b215c527-5737-4b21-a49e-8dc683225672>
  == Spawn extension (context_tok, 52431824, 2) exited non-zero on 'SIP/inbound_1_1-00000003'
Scheduling destruction of SIP dialog '5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp' in 128640 ms (Method: ACK)
Reliably Transmitting (NAT) to 61.213.230.153:5060:
BYE sip:05052424142@61.213.230.153:5060 SIP/2.0
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK7f6bdd13;rport
Max-Forwards: 70
From: <sip:52431824@52.194.253.25:15060>;tag=as303506f7
To: "05052424142" <sip:05052424142@smart.0038.net>;tag=as7d255076
Call-ID: 5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.11.0-rc1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:61.213.230.153:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 52.194.253.25:15060;branch=z9hG4bK7f6bdd13;received=52.194.253.25;rport=15060
From: <sip:52431824@52.194.253.25:15060>;tag=as303506f7
To: "05052424142" <sip:05052424142@smart.0038.net>;tag=as7d255076
Call-ID: 5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp
CSeq: 102 BYE
Server: Fusion Open MGW 1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '5b864ee94d613c180cf90d656fd1439e@fkr16.fusioncom.co.jp' Method: ACK

<--- SIP read from UDP:111.108.30.208:60645 --->


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

Update;

Register an account works without also any errors

xweezy · April 12, 2019, 2:15am

It seems, NAT problem, is your eth interface address a private IP address?

If it is, make sure you are using right network address in localnet.

If it is not, turn on the NAT option for force_rport,comedia in SIP extensions.

I hope all the required ports are open from the firewall.

wesly1988 · April 12, 2019, 2:29am

@xweezy, thank you for your fast reply.
It is highly appreciated.

The eth interface address is global.
To be honest we are already using the “force_rport,comedia” option for the sip extensions.

This is set in the general section and in the database.
All the ports are also standing open.

That’s what makes it quite strange.

xweezy · April 12, 2019, 2:33am

they why is it localnet set for private network address in the general section. You can remove it and try it once.

wesly1988 · April 12, 2019, 2:36am

I was thinking that also, but when I don’t use the localnet, it return to no audio at all.

wesly1988 · April 12, 2019, 2:44am

To make it a bit better, this is our database settings for the users

Sip user 1_1_1_1

mysql> select * from sip_clients limit 1 \G
*************************** 1. row ***************************
                id: 1
              name: 1_1_1_1
            ipaddr: 111.108.30.208
              port: 60463
        regseconds: 1555036926
       defaultuser: 1_1_1_1
       fullcontact: sip:1_1_1_1@111.108.30.208:60463^3Btransport=UDP^3Brinstance=6bfd99e76ad04f9d
         regserver: 
         useragent: Z 5.2.25 rv2.8.112
            lastms: 0
              host: dynamic
              type: friend
           context: context_tok
            permit: NULL
              deny: NULL
            secret: password
         md5secret: NULL
      remotesecret: NULL
         transport: NULL
          dtmfmode: rfc2833
       directmedia: NULL
               nat: force_rport,comedia
         callgroup: NULL
       pickupgroup: NULL
          language: NULL
             allow: NULL
          disallow: NULL
          insecure: NULL
         trustrpid: NULL
    progressinband: NULL
      promiscredir: NULL
     useclientcode: NULL
       accountcode: NULL
            setvar: NULL
          callerid: NULL
          amaflags: NULL
       callcounter: NULL
         busylevel: NULL
      allowoverlap: NULL
    allowsubscribe: NULL
      videosupport: NULL
    maxcallbitrate: NULL
 rfc2833compensate: NULL
           mailbox: NULL
    session-timers: NULL
   session-expires: NULL
     session-minse: NULL
 session-refresher: NULL
t38pt_usertpsource: NULL
          regexten: NULL
        fromdomain: NULL
          fromuser: NULL
           qualify: NULL
         defaultip: NULL
        rtptimeout: NULL
    rtpholdtimeout: NULL
          sendrpid: NULL
     outboundproxy: NULL
 callbackextension: NULL
           timert1: NULL
            timerb: NULL
       qualifyfreq: NULL
      constantssrc: NULL
     contactpermit: NULL
       contactdeny: NULL
       usereqphone: NULL
       textsupport: NULL
         faxdetect: NULL
          buggymwi: NULL
              auth: NULL
          fullname: NULL
         trunkname: NULL
        cid_number: NULL
       callingpres: NULL
      mohinterpret: NULL
        mohsuggest: NULL
        parkinglot: NULL
      hasvoicemail: NULL
      subscribemwi: NULL
           vmexten: NULL
       autoframing: NULL
      rtpkeepalive: NULL
        call-limit: 1
   g726nonstandard: NULL
  ignoresdpversion: NULL
     allowtransfer: NULL
           dynamic: NULL
1 row in set (0.00 sec)

Sip user 1_1_1_3

*************************** 2. row ***************************
                id: 3
              name: 1_1_1_3
            ipaddr: 111.108.30.208
              port: 60645
        regseconds: 1555036980
       defaultuser: 1_1_1_3
       fullcontact: sip:1_1_1_3@111.108.30.208:60645^3Btransport=UDP^3Brinstance=3c89da6bfdc8ddc8
         regserver: 
         useragent: Z 5.2.28 rv2.8.115
            lastms: 0
              host: dynamic
              type: friend
           context: context_tok
            permit: NULL
              deny: NULL
            secret: password
         md5secret: NULL
      remotesecret: NULL
         transport: NULL
          dtmfmode: rfc2833
       directmedia: NULL
               nat: force_rport,comedia
         callgroup: NULL
       pickupgroup: NULL
          language: NULL
             allow: NULL
          disallow: NULL
          insecure: NULL
         trustrpid: NULL
    progressinband: NULL
      promiscredir: NULL
     useclientcode: NULL
       accountcode: NULL
            setvar: NULL
          callerid: NULL
          amaflags: NULL
       callcounter: NULL
         busylevel: NULL
      allowoverlap: NULL
    allowsubscribe: NULL
      videosupport: NULL
    maxcallbitrate: NULL
 rfc2833compensate: NULL
           mailbox: NULL
    session-timers: NULL
   session-expires: NULL
     session-minse: NULL
 session-refresher: NULL
t38pt_usertpsource: NULL
          regexten: NULL
        fromdomain: NULL
          fromuser: NULL
           qualify: NULL
         defaultip: NULL
        rtptimeout: NULL
    rtpholdtimeout: NULL
          sendrpid: NULL
     outboundproxy: NULL
 callbackextension: NULL
           timert1: NULL
            timerb: NULL
       qualifyfreq: NULL
      constantssrc: NULL
     contactpermit: NULL
       contactdeny: NULL
       usereqphone: NULL
       textsupport: NULL
         faxdetect: NULL
          buggymwi: NULL
              auth: NULL
          fullname: NULL
         trunkname: NULL
        cid_number: NULL
       callingpres: NULL
      mohinterpret: NULL
        mohsuggest: NULL
        parkinglot: NULL
      hasvoicemail: NULL
      subscribemwi: NULL
           vmexten: NULL
       autoframing: NULL
      rtpkeepalive: NULL
        call-limit: 1
   g726nonstandard: NULL
  ignoresdpversion: NULL
     allowtransfer: NULL
           dynamic: NULL
2 rows in set (0.00 sec)

xweezy · April 12, 2019, 2:48am

you might need to add some default values in the fields. directmedia and codecs are not selected.

wesly1988 · April 12, 2019, 5:26am

Thank you. The problem is fixed.
I needed to put the qualify to yes and not null.

This made it work.
Thank you for your help!

ambiorixg12 · April 12, 2019, 4:18pm

Asterisk documentation clearly says that beside the NAT configuration also you need to turn on qualify=yes to keep the nat session open

wesly1988 · April 14, 2019, 11:47pm

Thank you @ambiorixg12,
To be honest, I had set it on the general section, this was because I had read it in the documentation.
I was just stupid to not check the database.

Smart lessons for next time.

system · May 14, 2019, 11:47pm

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