INBOUND CALLING NOT WORKING IN GOAUTODIAL

[wjohnson133]

I cannot, for the life of me get my phone to ring with inbound calling, I am now at the point where I am getting an echotest but nothing else, I have done show sip peers and debugging for my providers ip address, please let me know if you need anything else. I could not find where I could go to get all of the system information on CENTOS.

go*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
8001/8001 192.168.0.143 D N 43276 OK (1 ms)
8002/8002 (Unspecified) D N 0 UNKNOWN
8003/8003 (Unspecified) D N 0 UNKNOWN
8004/8004 (Unspecified) D N 0 UNKNOWN
8005/8005 (Unspecified) D N 0 UNKNOWN
8006/8006 (Unspecified) D N 0 UNKNOWN
8007/8007 (Unspecified) D N 0 UNKNOWN
8008/8008 (Unspecified) D N 0 UNKNOWN
8009/8009 (Unspecified) D N 0 UNKNOWN
8010/8010 (Unspecified) D N 0 UNKNOWN
8011/8011 (Unspecified) D N 0 UNKNOWN
8012/8012 (Unspecified) D N 0 UNKNOWN
8013/8013 (Unspecified) D N 0 UNKNOWN
8014/8014 (Unspecified) D N 0 UNKNOWN
8015/8015 (Unspecified) D N 0 UNKNOWN
8016/8016 (Unspecified) D N 0 UNKNOWN
8017/8017 (Unspecified) D N 0 UNKNOWN
8018/8018 (Unspecified) D N 0 UNKNOWN
8019/8019 (Unspecified) D N 0 UNKNOWN
8020/8020 (Unspecified) D N 0 UNKNOWN
8050 (Unspecified) D N 0 UNKNOWN
MYOWNTELCO2/xxxxx xxx.204.72.125 N 5060 OK (13 ms)
MYOWNTELCO3/xxxxx xxx.204.72.125 5060 OK (15 ms)
MyOwnTelco/xxxxx xxx.204.72.125 N 5060 OK (17 ms)
24 sip peers [Monitored: 4 online, 20 offline Unmonitored: 0 online, 0 offline]
[Mar 28 14:47:01] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 28 14:47:01] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 28 14:47:01] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 28 14:47:03] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 28 14:47:06] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 28 14:47:06] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 28 14:48:02] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 28 14:48:02] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 28 14:48:02] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 28 14:48:03] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 28 14:48:07] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 28 14:48:07] == Manager 'sendcron' logged off from 127.0.0.1
go*CLI> sip set debug ip xxx.204.72.125
SIP Debugging Enabled for IP: xxx.204.72.125
[Mar 28 14:48:24]
<--- SIP read from UDP:xxx.204.72.125:5060 --->
INVITE sip:xxxxx@192.168.0.12:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.204.72.125:5060;branch=z9hG4bK4d5a90e6;rport
Max-Forwards: 70
From: "5144647297" <sip:5144647297@xxx.204.72.125>;tag=as70354a48
To: <sip:xxxxx@192.168.0.12:5060>
Contact: <sip:5144647297@xxx.204.72.125:5060>
Call-ID: 6163a8121b8974c40cf85eef661524db@xxx.204.72.125:5060
CSeq: 102 INVITE
User-Agent: MyOwnAccount
Date: Tue, 28 Mar 2017 18:48:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 336

v=0
o=root 721627889 721627889 IN IP4 xxx.204.72.125
s=Asterisk PBX 1.8.24.0
c=IN IP4 xxx.204.72.125
t=0 0
m=audio 13670 RTP/AVP 0 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
[Mar 28 14:48:24] --- (14 headers 15 lines) ---
[Mar 28 14:48:24] Sending to xxx.204.72.125:5060 (NAT)
[Mar 28 14:48:24] Using INVITE request as basis request - 6163a8121b8974c40cf85eef661524db@xxx.204.72.125:5060
[Mar 28 14:48:24] Found peer 'MyOwnTelco' for '5144647297' from xxx.204.72.125:5060
[Mar 28 14:48:24] == Using SIP RTP CoS mark 5
[Mar 28 14:48:24] Found RTP audio format 0
[Mar 28 14:48:24] Found RTP audio format 18
[Mar 28 14:48:24] Found RTP audio format 3
[Mar 28 14:48:24] Found RTP audio format 101
[Mar 28 14:48:24] Found audio description format PCMU for ID 0
[Mar 28 14:48:24] Found audio description format G729 for ID 18
[Mar 28 14:48:24] Found audio description format GSM for ID 3
[Mar 28 14:48:24] Found audio description format telephone-event for ID 101
[Mar 28 14:48:24] Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x106 (gsm|ulaw|g729)
[Mar 28 14:48:24] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Mar 28 14:48:24] Peer audio RTP is at port xxx.204.72.125:13670
[Mar 28 14:48:24] Looking for xxxxx in trunkinbound (domain 192.168.0.12)
[Mar 28 14:48:24] list_route: hop: <sip:5144647297@xxx.204.72.125:5060>
[Mar 28 14:48:24]
<--- Transmitting (NAT) to xxx.204.72.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.204.72.125:5060;branch=z9hG4bK4d5a90e6;received=xxx.204.72.125;rport=5060
From: "5144647297" <sip:5144647297@xxx.204.72.125>;tag=as70354a48
To: <sip:xxxxx@192.168.0.12:5060>
Call-ID: 6163a8121b8974c40cf85eef661524db@xxx.204.72.125:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:xxxxx@192.168.0.12:5060>
Content-Length: 0


<------------>
[Mar 28 14:48:24] -- Executing [xxxxx@trunkinbound:1] Answer("SIP/MyOwnTelco-00000007", "") in new stack
[Mar 28 14:48:24] Audio is at 13540
[Mar 28 14:48:24] Adding codec 0x2 (gsm) to SDP
[Mar 28 14:48:24] Adding codec 0x4 (ulaw) to SDP
[Mar 28 14:48:24] Adding codec 0x100 (g729) to SDP
[Mar 28 14:48:24] Adding non-codec 0x1 (telephone-event) to SDP
[Mar 28 14:48:24]
<--- Reliably Transmitting (NAT) to xxx.204.72.125:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.204.72.125:5060;branch=z9hG4bK4d5a90e6;received=xxx.204.72.125;rport=5060
From: "5144647297" <sip:5144647297@xxx.204.72.125>;tag=as70354a48
To: <sip:xxxxx@192.168.0.12:5060>;tag=as1b738653
Call-ID: 6163a8121b8974c40cf85eef661524db@xxx.204.72.125:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:xxxxx@192.168.0.12:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 347

v=0
o=root 444993535 444993535 IN IP4 192.168.0.12
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 192.168.0.12
t=0 0
m=audio 13540 RTP/AVP 3 0 18 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Mar 28 14:48:24]
<--- SIP read from UDP:xxx.204.72.125:5060 --->
ACK sip:xxxxx@192.168.0.12:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.204.72.125:5060;branch=z9hG4bK13470a13;rport
Max-Forwards: 70
From: "5144647297" <sip:5144647297@xxx.204.72.125>;tag=as70354a48
To: <sip:xxxxx@192.168.0.12:5060>;tag=as1b738653
Contact: <sip:5144647297@xxx.204.72.125:5060>
Call-ID: 6163a8121b8974c40cf85eef661524db@xxx.204.72.125:5060
CSeq: 102 ACK
User-Agent: MyOwnAccount
Content-Length: 0

<------------->
[Mar 28 14:48:24] --- (10 headers 0 lines) ---
[Mar 28 14:48:24] -- Executing [xxxxx@trunkinbound:2] Playback("SIP/MyOwnTelco-00000007", "demo-echotest") in new stack
[Mar 28 14:48:24] -- <SIP/MyOwnTelco-00000007> Playing 'demo-echotest.gsm' (language 'en')
[Mar 28 14:48:24] NOTICE[9217]: channel.c:4237 __ast_read: Dropping incompatible voice frame on SIP/MyOwnTelco-00000007 of format gsm since our native format has changed to 0x4 (ulaw)
[Mar 28 14:48:44] -- Executing [xxxxx@trunkinbound:3] Hangup("SIP/MyOwnTelco-00000007", "") in new stack
[Mar 28 14:48:44] == Spawn extension (trunkinbound, xxxxx, 3) exited non-zero on 'SIP/MyOwnTelco-00000007'
[Mar 28 14:48:44] -- Executing [h@trunkinbound:1] AGI("SIP/MyOwnTelco-00000007", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Mar 28 14:48:44] -- <SIP/MyOwnTelco-00000007>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 28 14:48:44] -- Executing [h@trunkinbound:2] Playback("SIP/MyOwnTelco-00000007", "demo-echotest") in new stack
[Mar 28 14:48:44] WARNING[9217]: file.c:767 ast_readaudio_callback: Failed to write frame
[Mar 28 14:48:44] -- <SIP/MyOwnTelco-00000007> Playing 'demo-echotest.gsm' (language 'en')
[Mar 28 14:48:44] WARNING[9217]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/MyOwnTelco-00000007 for demo-echotest
[Mar 28 14:48:44] -- Executing [h@trunkinbound:3] Hangup("SIP/MyOwnTelco-00000007", "") in new stack
[Mar 28 14:48:44] == Spawn extension (trunkinbound, h, 3) exited non-zero on 'SIP/MyOwnTelco-00000007'
[Mar 28 14:48:44] Scheduling destruction of SIP dialog '6163a8121b8974c40cf85eef661524db@xxx.204.72.125:5060' in 6400 ms (Method: ACK)
[Mar 28 14:48:44] set_destination: Parsing <sip:5144647297@xxx.204.72.125:5060> for address/port to send to
[Mar 28 14:48:44] set_destination: set destination to xxx.204.72.125:5060
[Mar 28 14:48:44] Reliably Transmitting (NAT) to xxx.204.72.125:5060:
BYE sip:5144647297@xxx.204.72.125:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK26c8d647;rport
Max-Forwards: 70
From: <sip:xxxxx@192.168.0.12:5060>;tag=as1b738653
To: "5144647297" <sip:5144647297@xxx.204.72.125>;tag=as70354a48
Call-ID: 6163a8121b8974c40cf85eef661524db@xxx.204.72.125:5060
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0