No audio and disconnect after 2secs

hi,

Need help can’t dial out.

logs

REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK43274e0f;rport
Max-Forwards: 70
From: sip:[email protected];tag=as6932a324
To: sip:[email protected]
Call-ID: [email protected][::1]
CSeq: 114 REGISTER
User-Agent: FPBX-2.10.1(1.8.25.0)
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0


REGISTER 10 headers, 0 lines
Reliably Transmitting (NAT) to 64.34.181.47:5060:
REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK67ec288e;rport
Max-Forwards: 70
From: sip:[email protected];tag=as53ec10d0
To: sip:[email protected]
Call-ID: [email protected][::1]
CSeq: 115 REGISTER
User-Agent: FPBX-2.10.1(1.8.25.0)
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0


Really destroying SIP dialog ‘[email protected][::1]’ Method: REGISTER
Retransmitting #1 (NAT) to 64.34.181.47:5060:
REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK67ec288e;rport
Max-Forwards: 70
From: sip:[email protected];tag=as53ec10d0
To: sip:[email protected]
Call-ID: [email protected][::1]
CSeq: 115 REGISTER
User-Agent: FPBX-2.10.1(1.8.25.0)
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0


Reliably Transmitting (NAT) to X.X.X.30:53796:
OPTIONS sip:[email protected]:9043 SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK49678599;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as7efc395d
To: sip:[email protected]:9043
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.25.0)
Date: Mon, 28 Apr 2014 20:23:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:X.X.X.30:53796 —>
SIP/2.0 200 OK
To: sip:[email protected]:9043;tag=8074730c
From: "Unknown"sip:[email protected];tag=as7efc395d
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK49678599;rport=5060;received=X.X.X.34
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Contact: sip:192.168.1.84:9043
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: eventlist
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

<— SIP read from UDP:208.74.75.250:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 202.78.76.150:5060;received=X.X.X.34;branch=z9hG4bK4ce0ca04;rport=5060
Record-Route: sip:[email protected]:5060;lr;transport=udp
To: sip:[email protected];tag=sansay1901945923rdb26441
From: “7025532722” sip:[email protected];tag=as7614f601
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 277

v=0
o=Sansay-VSXi 188 1 IN IP4 208.74.75.250
s=Session Controller
c=IN IP4 208.74.75.252
t=0 0
m=audio 24996 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
<------------->
— (10 headers 13 lines) —
list_route: hop: sip:[email protected]:5060;lr;transport=udp
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 208.74.75.252:24996
– SIP/apnout-00000036 is making progress passing it to SIP/200-00000035
Audio is at 17996
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to X.X.X.30:53796 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.84:9043;branch=z9hG4bK-d87543-1026317938-1–d87543-;received=X.X.X.30;rport=53796
From: sip:[email protected];tag=756e4274
To: sip:[email protected];tag=as066d3f67
Call-ID: e60ecc2fe35e682f
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1820299014 1820299014 IN IP4 202.78.76.150
s=Asterisk PBX 1.8.25.0
c=IN IP4 202.78.76.150
t=0 0
m=audio 17996 RTP/AVP 18 8 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Retransmitting #2 (NAT) to 64.34.181.47:5060:
REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK67ec288e;rport
Max-Forwards: 70
From: sip:[email protected];tag=as53ec10d0
To: sip:[email protected]
Call-ID: [email protected][::1]
CSeq: 115 REGISTER
User-Agent: FPBX-2.10.1(1.8.25.0)
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0


<— SIP read from UDP:208.74.75.250:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.78.76.150:5060;received=X.X.X.34;branch=z9hG4bK4ce0ca04;rport=5060
Record-Route: sip:[email protected]:5060;lr;transport=udp
To: sip:[email protected];tag=sansay1901945923rdb26441
From: “7025532722” sip:[email protected];tag=as7614f601
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 277

v=0
o=Sansay-VSXi 188 1 IN IP4 208.74.75.250
s=Session Controller
c=IN IP4 208.74.75.252
t=0 0
m=audio 24996 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
<------------->
— (10 headers 13 lines) —
list_route: hop: sip:[email protected]:5060;lr;transport=udp
set_destination: Parsing sip:[email protected]:5060;lr;transport=udp for address/port to send to
set_destination: set destination to 208.74.75.250:5060
Transmitting (NAT) to 208.74.75.250:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK38a430f2;rport
Route: sip:[email protected]:5060;lr;transport=udp
Max-Forwards: 70
From: “7025532722” sip:[email protected];tag=as7614f601
To: sip:[email protected];tag=sansay1901945923rdb26441
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: FPBX-2.10.1(1.8.25.0)
Content-Length: 0


-- SIP/apnout-00000036 answered SIP/200-00000035

Audio is at 17996
Adding codec 0x100 (g729) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to X.X.X.30:53796 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:9043;branch=z9hG4bK-d87543-1026317938-1–d87543-;received=X.X.X.30;rport=53796
From: sip:[email protected];tag=756e4274
To: sip:[email protected];tag=as066d3f67
Call-ID: e60ecc2fe35e682f
Seq: 2 INVITE
Server: FPBX-2.10.1(1.8.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1820299014 1820299014 IN IP4 202.78.76.150
s=Asterisk PBX 1.8.25.0
c=IN IP4 202.78.76.150
t=0 0
m=audio 17996 RTP/AVP 18 8 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Retransmitting #1 (NAT) to X.X.X.30:53796:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:9043;branch=z9hG4bK-d87543-1026317938-1–d87543-;received=X.X.X.30;rport=53796
From: sip:[email protected];tag=756e4274
To: sip:[email protected];tag=as066d3f67
Call-ID: e60ecc2fe35e682f
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1820299014 1820299014 IN IP4 202.78.76.150
s=Asterisk PBX 1.8.25.0
c=IN IP4 202.78.76.150
t=0 0
m=audio 17996 RTP/AVP 18 8 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


Retransmitting #2 (NAT) to X.X.X.30:53796:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:9043;branch=z9hG4bK-d87543-1026317938-1–d87543-;received=X.X.X.30;rport=53796
From: sip:[email protected];tag=756e4274
To: sip:[email protected];tag=as066d3f67
Call-ID: e60ecc2fe35e682f
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1820299014 1820299014 IN IP4 202.78.76.150
s=Asterisk PBX 1.8.25.0
c=IN IP4 202.78.76.150
t=0 0
m=audio 17996 RTP/AVP 18 8 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


Retransmitting #3 (NAT) to X.X.X.30:53796:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:9043;branch=z9hG4bK-d87543-1026317938-1–d87543-;received=X.X.X.30;rport=53796
From: sip:[email protected];tag=756e4274
To: sip:[email protected];tag=as066d3f67
Call-ID: e60ecc2fe35e682f
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1820299014 1820299014 IN IP4 202.78.76.150
s=Asterisk PBX 1.8.25.0
c=IN IP4 202.78.76.150
t=0 0
m=audio 17996 RTP/AVP 18 8 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


<— SIP read from UDP:X.X.X.30:53796 —>

<------------->
Retransmitting #4 (NAT) to X.X.X.30:53796:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:9043;branch=z9hG4bK-d87543-1026317938-1–d87543-;received=X.X.X.30;rport=53796
From: sip:[email protected];tag=756e4274
To: sip:[email protected];tag=as066d3f67
Call-ID: e60ecc2fe35e682f
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1820299014 1820299014 IN IP4 202.78.76.150
s=Asterisk PBX 1.8.25.0
c=IN IP4 202.78.76.150
t=0 0
m=audio 17996 RTP/AVP 18 8 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


Retransmitting #3 (NAT) to 64.34.181.47:5060:
REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK67ec288e;rport
Max-Forwards: 70
From: sip:[email protected];tag=as53ec10d0
To: sip:[email protected]
Call-ID: [email protected][::1]
CSeq: 115 REGISTER
User-Agent: FPBX-2.10.1(1.8.25.0)
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0


Retransmitting #5 (NAT) to X.X.X.30:53796:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.84:9043;branch=z9hG4bK-d87543-1026317938-1–d87543-;received=X.X.X.30;rport=53796
From: sip:[email protected];tag=756e4274
To: sip:[email protected];tag=as066d3f67
Call-ID: e60ecc2fe35e682f
CSeq: 2 INVITE
Server: FPBX-2.10.1(1.8.25.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 1820299014 1820299014 IN IP4 202.78.76.150
s=Asterisk PBX 1.8.25.0
c=IN IP4 202.78.76.150
t=0 0
m=audio 17996 RTP/AVP 18 8 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


Reliably Transmitting (no NAT) to 192.168.1.59:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK466b5968
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as090d7758
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.25.0)
Date: Mon, 28 Apr 2014 20:23:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #1 (no NAT) to 192.168.1.59:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK466b5968
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as090d7758
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.25.0)
Date: Mon, 28 Apr 2014 20:23:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #2 (no NAT) to 192.168.1.59:5060:
OPTIONS sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK466b5968
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as090d7758
To: sip:[email protected]:5060;transport=udp
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.25.0)
Date: Mon, 28 Apr 2014 20:23:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


Retransmitting #4 (NAT) to 64.34.181.47:5060:
REGISTER sip:did.voip.les.net SIP/2.0
Via: SIP/2.0/UDP 202.78.76.150:5060;branch=z9hG4bK67ec288e;rport
Max-Forwards: 70
From: sip:[email protected];tag=as53ec10d0
To: sip:[email protected]
Call-ID: [email protected][::1]
CSeq: 115 REGISTER
User-Agent: FPBX-2.10.1(1.8.25.0)
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0