10 sec of delay on sip call (debug log attached)

Hi,
I hope you may help me. I have strange 10 sec of dellay on all sip calls (internal, external, incoming, outgoing)

log for incoming call from trunk attached. you can see on the timestamp about 10 sec on delay from the initial invite to the end of the handshake.
any idea ?
Thanks



localhost*CLI>
[Sep  8 17:16:12] Really destroying SIP dialog '2080916178_73533615@211.184.214.21' Method: BYE
[Sep  8 17:16:13]
[Sep  8 17:16:13] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:13] INVITE sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:13] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e
[Sep  8 17:16:13] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:13] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:13] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:13] CSeq: 20717 INVITE
[Sep  8 17:16:13] Max-Forwards: 68
[Sep  8 17:16:13] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:13] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:13] Contact: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:13] P-Asserted-Identity: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:13] Supported: timer,100rel
[Sep  8 17:16:13] Session-Expires: 1800
[Sep  8 17:16:13] Min-SE: 90
[Sep  8 17:16:13] Content-Length: 285
[Sep  8 17:16:13] Content-Disposition: session; handling=required
[Sep  8 17:16:13] Content-Type: application/sdp
[Sep  8 17:16:13]
[Sep  8 17:16:13] v=0
[Sep  8 17:16:13] o=Sonus_UAC 1438 27200 IN IP4 211.184.214.21
[Sep  8 17:16:13] s=SIP Media Capabilities
[Sep  8 17:16:13] c=IN IP4 212.199.220.22
[Sep  8 17:16:13] t=0 0
[Sep  8 17:16:13] m=audio 6524 RTP/AVP 8 18 100
[Sep  8 17:16:13] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:13] a=rtpmap:18 G729/8000
[Sep  8 17:16:13] a=fmtp:18 annexb=no
[Sep  8 17:16:13] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:13] a=fmtp:100 0-15
[Sep  8 17:16:13] a=sendrecv
[Sep  8 17:16:13] a=maxptime:20
[Sep  8 17:16:13] <------------->
[Sep  8 17:16:13] --- (17 headers 13 lines) ---
[Sep  8 17:16:13] Sending to 211.184.214.21:5060 (no NAT)
[Sep  8 17:16:13] Sending to 211.184.214.21:5060 (no NAT)
[Sep  8 17:16:13] Using INVITE request as basis request - 2080916219_57685074@211.184.214.21
[Sep  8 17:16:13] Found peer '012' for '0578458456' from 211.184.214.21:5060
[Sep  8 17:16:22] Found RTP audio format 8
[Sep  8 17:16:22] Found RTP audio format 18
[Sep  8 17:16:22] Found RTP audio format 100
[Sep  8 17:16:22] Found audio description format PCMA for ID 8
[Sep  8 17:16:22] Found audio description format G729 for ID 18
[Sep  8 17:16:22] Found audio description format telephone-event for ID 100
[Sep  8 17:16:22] Capabilities: us - (ulaw|alaw|gsm|g722|g729|g723), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729)
[Sep  8 17:16:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Sep  8 17:16:22] Peer audio RTP is at port 212.199.220.22:6524
[Sep  8 17:16:22] Looking for 722798275 in from-trunk-sip-012 (domain 212.68.136.179)
[Sep  8 17:16:22] sip_route_dump: route/path hop: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 100 Trying
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22] Reliably Transmitting (no NAT) to 211.184.214.21:5060:
[Sep  8 17:16:22] OPTIONS sip:211.184.214.21 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 212.68.136.179:5060;branch=z9hG4bK13789235
[Sep  8 17:16:22] Max-Forwards: 70
[Sep  8 17:16:22] From: "Unknown" <sip:Unknown@212.68.136.179>;tag=as6a0deba7
[Sep  8 17:16:22] To: <sip:211.184.214.21>
[Sep  8 17:16:22] Contact: <sip:Unknown@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 03565ee7412794ab49e29b3c4d89b0f6@212.68.136.179:5060
[Sep  8 17:16:22] CSeq: 102 OPTIONS
[Sep  8 17:16:22] User-Agent: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Date: Tue, 08 Sep 2015 14:16:22 GMT
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] ---
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] INVITE sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Max-Forwards: 68
[Sep  8 17:16:22] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:22] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:22] Contact: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] P-Asserted-Identity: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] Supported: timer,100rel
[Sep  8 17:16:22] Session-Expires: 1800
[Sep  8 17:16:22] Min-SE: 90
[Sep  8 17:16:22] Content-Length: 285
[Sep  8 17:16:22] Content-Disposition: session; handling=required
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=Sonus_UAC 1438 27200 IN IP4 211.184.214.21
[Sep  8 17:16:22] s=SIP Media Capabilities
[Sep  8 17:16:22] c=IN IP4 212.199.220.22
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 6524 RTP/AVP 8 18 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-15
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22] a=maxptime:20
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (17 headers 13 lines) ---
[Sep  8 17:16:22] Ignoring this INVITE request
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 100 Trying
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] INVITE sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Max-Forwards: 68
[Sep  8 17:16:22] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:22] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:22] Contact: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] P-Asserted-Identity: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] Supported: timer,100rel
[Sep  8 17:16:22] Session-Expires: 1800
[Sep  8 17:16:22] Min-SE: 90
[Sep  8 17:16:22] Content-Length: 285
[Sep  8 17:16:22] Content-Disposition: session; handling=required
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=Sonus_UAC 1438 27200 IN IP4 211.184.214.21
[Sep  8 17:16:22] s=SIP Media Capabilities
[Sep  8 17:16:22] c=IN IP4 212.199.220.22
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 6524 RTP/AVP 8 18 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-15
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22] a=maxptime:20
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (17 headers 13 lines) ---
[Sep  8 17:16:22] Ignoring this INVITE request
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 100 Trying
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] INVITE sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Max-Forwards: 68
[Sep  8 17:16:22] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:22] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:22] Contact: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] P-Asserted-Identity: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] Supported: timer,100rel
[Sep  8 17:16:22] Session-Expires: 1800
[Sep  8 17:16:22] Min-SE: 90
[Sep  8 17:16:22] Content-Length: 285
[Sep  8 17:16:22] Content-Disposition: session; handling=required
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=Sonus_UAC 1438 27200 IN IP4 211.184.214.21
[Sep  8 17:16:22] s=SIP Media Capabilities
[Sep  8 17:16:22] c=IN IP4 212.199.220.22
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 6524 RTP/AVP 8 18 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-15
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22] a=maxptime:20
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (17 headers 13 lines) ---
[Sep  8 17:16:22] Ignoring this INVITE request
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 100 Trying
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] INVITE sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Max-Forwards: 68
[Sep  8 17:16:22] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:22] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:22] Contact: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] P-Asserted-Identity: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] Supported: timer,100rel
[Sep  8 17:16:22] Session-Expires: 1800
[Sep  8 17:16:22] Min-SE: 90
[Sep  8 17:16:22] Content-Length: 285
[Sep  8 17:16:22] Content-Disposition: session; handling=required
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=Sonus_UAC 1438 27200 IN IP4 211.184.214.21
[Sep  8 17:16:22] s=SIP Media Capabilities
[Sep  8 17:16:22] c=IN IP4 212.199.220.22
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 6524 RTP/AVP 8 18 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-15
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22] a=maxptime:20
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (17 headers 13 lines) ---
[Sep  8 17:16:22] Ignoring this INVITE request
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 100 Trying
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[2015-09-08 17:16:22] WARNING[23211][C-000006e0]: func_channel.c:596 func_channel_read: Unknown or unavailable item requested: 'reversecharge'
[Sep  8 17:16:22] Audio is at 15346
[Sep  8 17:16:22] Adding codec alaw to SDP
[Sep  8 17:16:22] Adding codec g729 to SDP
[Sep  8 17:16:22] Adding codec ulaw to SDP
[Sep  8 17:16:22] Adding codec gsm to SDP
[Sep  8 17:16:22] Adding codec g722 to SDP
[Sep  8 17:16:22] Adding codec g723 to SDP
[Sep  8 17:16:22] Adding non-codec 0x1 (telephone-event) to SDP
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Reliably Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 200 OK
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22] Require: timer
[Sep  8 17:16:22] Content-Length: 415
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=root 579383943 579383943 IN IP4 212.68.136.179
[Sep  8 17:16:22] s=Asterisk PBX 13.4.0
[Sep  8 17:16:22] c=IN IP4 212.68.136.179
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 15346 RTP/AVP 8 18 0 3 9 4 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:0 PCMU/8000
[Sep  8 17:16:22] a=rtpmap:3 GSM/8000
[Sep  8 17:16:22] a=rtpmap:9 G722/8000
[Sep  8 17:16:22] a=rtpmap:4 G723/8000
[Sep  8 17:16:22] a=fmtp:4 annexa=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-16
[Sep  8 17:16:22] a=ptime:20
[Sep  8 17:16:22] a=maxptime:150
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 200 OK
[Sep  8 17:16:22] Via: SIP/2.0/UDP 212.68.136.179:5060;branch=z9hG4bK13789235
[Sep  8 17:16:22] From: "Unknown" <sip:Unknown@212.68.136.179>;tag=as6a0deba7
[Sep  8 17:16:22] To: <sip:211.184.214.21>
[Sep  8 17:16:22] Call-ID: 03565ee7412794ab49e29b3c4d89b0f6@212.68.136.179:5060
[Sep  8 17:16:22] CSeq: 102 OPTIONS
[Sep  8 17:16:22] Contact: <sip:ANONYMOUS@211.184.214.21:5060>
[Sep  8 17:16:22] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:22] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:22] Supported: timer,100rel,replaces
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (11 headers 0 lines) ---
[Sep  8 17:16:22] Really destroying SIP dialog '03565ee7412794ab49e29b3c4d89b0f6@212.68.136.179:5060' Method: OPTIONS
[Sep  8 17:16:22] Retransmitting #1 (no NAT) to 211.184.214.21:5060:
[Sep  8 17:16:22] SIP/2.0 200 OK
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B0609768e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22] Require: timer
[Sep  8 17:16:22] Content-Length: 415
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=root 579383943 579383943 IN IP4 212.68.136.179
[Sep  8 17:16:22] s=Asterisk PBX 13.4.0
[Sep  8 17:16:22] c=IN IP4 212.68.136.179
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 15346 RTP/AVP 8 18 0 3 9 4 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:0 PCMU/8000
[Sep  8 17:16:22] a=rtpmap:3 GSM/8000
[Sep  8 17:16:22] a=rtpmap:9 G722/8000
[Sep  8 17:16:22] a=rtpmap:4 G723/8000
[Sep  8 17:16:22] a=fmtp:4 annexa=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-16
[Sep  8 17:16:22] a=ptime:20
[Sep  8 17:16:22] a=maxptime:150
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22]
[Sep  8 17:16:22] ---
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] ACK sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06113da296e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20717 ACK
[Sep  8 17:16:22] Max-Forwards: 70
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (8 headers 0 lines) ---
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] INVITE sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06121b9e96e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20718 INVITE
[Sep  8 17:16:22] Max-Forwards: 70
[Sep  8 17:16:22] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:22] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:22] Contact: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] Supported: timer,100rel
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Min-SE: 90
[Sep  8 17:16:22] Content-Length: 238
[Sep  8 17:16:22] Content-Disposition: session; handling=required
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=Sonus_UAC 1438 27201 IN IP4 211.184.214.21
[Sep  8 17:16:22] s=SIP Media Capabilities
[Sep  8 17:16:22] c=IN IP4 212.199.220.22
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 6524 RTP/AVP 8 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-15
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22] a=maxptime:20
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (16 headers 11 lines) ---
[Sep  8 17:16:22] Sending to 211.184.214.21:5060 (no NAT)
[Sep  8 17:16:22] Found RTP audio format 8
[Sep  8 17:16:22] Found RTP audio format 100
[Sep  8 17:16:22] Found audio description format PCMA for ID 8
[Sep  8 17:16:22] Found audio description format telephone-event for ID 100
[Sep  8 17:16:22] Capabilities: us - (ulaw|alaw|gsm|g722|g729|g723), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Sep  8 17:16:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Sep  8 17:16:22] Peer audio RTP is at port 212.199.220.22:6524
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 100 Trying
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06121b9e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20718 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22] Audio is at 15346
[Sep  8 17:16:22] Adding codec alaw to SDP
[Sep  8 17:16:22] Adding codec ulaw to SDP
[Sep  8 17:16:22] Adding codec gsm to SDP
[Sep  8 17:16:22] Adding codec g722 to SDP
[Sep  8 17:16:22] Adding codec g729 to SDP
[Sep  8 17:16:22] Adding codec g723 to SDP
[Sep  8 17:16:22] Adding non-codec 0x1 (telephone-event) to SDP
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Reliably Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 200 OK
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06121b9e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20718 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22] Require: timer
[Sep  8 17:16:22] Content-Length: 415
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=root 579383943 579383944 IN IP4 212.68.136.179
[Sep  8 17:16:22] s=Asterisk PBX 13.4.0
[Sep  8 17:16:22] c=IN IP4 212.68.136.179
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 15346 RTP/AVP 8 0 3 9 18 4 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:0 PCMU/8000
[Sep  8 17:16:22] a=rtpmap:3 GSM/8000
[Sep  8 17:16:22] a=rtpmap:9 G722/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:4 G723/8000
[Sep  8 17:16:22] a=fmtp:4 annexa=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-16
[Sep  8 17:16:22] a=ptime:20
[Sep  8 17:16:22] a=maxptime:150
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22] Retransmitting #1 (no NAT) to 211.184.214.21:5060:
[Sep  8 17:16:22] SIP/2.0 200 OK
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06121b9e96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20718 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22] Require: timer
[Sep  8 17:16:22] Content-Length: 415
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=root 579383943 579383944 IN IP4 212.68.136.179
[Sep  8 17:16:22] s=Asterisk PBX 13.4.0
[Sep  8 17:16:22] c=IN IP4 212.68.136.179
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 15346 RTP/AVP 8 0 3 9 18 4 100
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:0 PCMU/8000
[Sep  8 17:16:22] a=rtpmap:3 GSM/8000
[Sep  8 17:16:22] a=rtpmap:9 G722/8000
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:4 G723/8000
[Sep  8 17:16:22] a=fmtp:4 annexa=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-16
[Sep  8 17:16:22] a=ptime:20
[Sep  8 17:16:22] a=maxptime:150
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22]
[Sep  8 17:16:22] ---
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] ACK sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06147f4296e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20718 ACK
[Sep  8 17:16:22] Max-Forwards: 70
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (8 headers 0 lines) ---
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] ACK sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06155afc96e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20718 ACK
[Sep  8 17:16:22] Max-Forwards: 70
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (8 headers 0 lines) ---
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] INVITE sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06166b3f96e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20719 INVITE
[Sep  8 17:16:22] Max-Forwards: 70
[Sep  8 17:16:22] Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS
[Sep  8 17:16:22] Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
[Sep  8 17:16:22] Contact: <sip:0578458456@211.184.214.21:5060>
[Sep  8 17:16:22] Supported: timer,100rel
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Min-SE: 90
[Sep  8 17:16:22] Content-Length: 261
[Sep  8 17:16:22] Content-Disposition: session; handling=required
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=Sonus_UAC 1438 27202 IN IP4 211.184.214.21
[Sep  8 17:16:22] s=SIP Media Capabilities
[Sep  8 17:16:22] c=IN IP4 212.199.220.22
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 6524 RTP/AVP 18 100
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-15
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22] a=maxptime:20
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (16 headers 12 lines) ---
[Sep  8 17:16:22] Sending to 211.184.214.21:5060 (no NAT)
[Sep  8 17:16:22] Found RTP audio format 18
[Sep  8 17:16:22] Found RTP audio format 100
[Sep  8 17:16:22] Found audio description format G729 for ID 18
[Sep  8 17:16:22] Found audio description format telephone-event for ID 100
[Sep  8 17:16:22] Capabilities: us - (ulaw|alaw|gsm|g722|g729|g723), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
[Sep  8 17:16:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Sep  8 17:16:22] Peer audio RTP is at port 212.199.220.22:6524
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 100 Trying
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06166b3f96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20719 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22] Audio is at 15346
[Sep  8 17:16:22] Adding codec g729 to SDP
[Sep  8 17:16:22] Adding codec ulaw to SDP
[Sep  8 17:16:22] Adding codec alaw to SDP
[Sep  8 17:16:22] Adding codec gsm to SDP
[Sep  8 17:16:22] Adding codec g722 to SDP
[Sep  8 17:16:22] Adding codec g723 to SDP
[Sep  8 17:16:22] Adding non-codec 0x1 (telephone-event) to SDP
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- Reliably Transmitting (no NAT) to 211.184.214.21:5060 --->
[Sep  8 17:16:22] SIP/2.0 200 OK
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06166b3f96e3508e;received=211.184.214.21
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20719 INVITE
[Sep  8 17:16:22] Server: FPBX-12.0.76(13.4.0)
[Sep  8 17:16:22] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Sep  8 17:16:22] Supported: replaces, timer
[Sep  8 17:16:22] Session-Expires: 1800;refresher=uas
[Sep  8 17:16:22] Contact: <sip:722798275@212.68.136.179:5060>
[Sep  8 17:16:22] Content-Type: application/sdp
[Sep  8 17:16:22] Require: timer
[Sep  8 17:16:22] Content-Length: 415
[Sep  8 17:16:22]
[Sep  8 17:16:22] v=0
[Sep  8 17:16:22] o=root 579383943 579383945 IN IP4 212.68.136.179
[Sep  8 17:16:22] s=Asterisk PBX 13.4.0
[Sep  8 17:16:22] c=IN IP4 212.68.136.179
[Sep  8 17:16:22] t=0 0
[Sep  8 17:16:22] m=audio 15346 RTP/AVP 18 0 8 3 9 4 100
[Sep  8 17:16:22] a=rtpmap:18 G729/8000
[Sep  8 17:16:22] a=fmtp:18 annexb=no
[Sep  8 17:16:22] a=rtpmap:0 PCMU/8000
[Sep  8 17:16:22] a=rtpmap:8 PCMA/8000
[Sep  8 17:16:22] a=rtpmap:3 GSM/8000
[Sep  8 17:16:22] a=rtpmap:9 G722/8000
[Sep  8 17:16:22] a=rtpmap:4 G723/8000
[Sep  8 17:16:22] a=fmtp:4 annexa=no
[Sep  8 17:16:22] a=rtpmap:100 telephone-event/8000
[Sep  8 17:16:22] a=fmtp:100 0-16
[Sep  8 17:16:22] a=ptime:20
[Sep  8 17:16:22] a=maxptime:150
[Sep  8 17:16:22] a=sendrecv
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------>
[Sep  8 17:16:22]
[Sep  8 17:16:22] <--- SIP read from UDP:211.184.214.21:5060 --->
[Sep  8 17:16:22] ACK sip:722798275@212.68.136.179:5060 SIP/2.0
[Sep  8 17:16:22] Via: SIP/2.0/UDP 211.184.214.21:5060;branch=z9hG4bK08B06172df896e3508e
[Sep  8 17:16:22] From: <sip:0578458456@211.184.214.21:5060>;tag=gK0863ed45
[Sep  8 17:16:22] To: <sip:722798275@212.68.136.179:5060>;tag=as0c5d8a1d
[Sep  8 17:16:22] Call-ID: 2080916219_57685074@211.184.214.21
[Sep  8 17:16:22] CSeq: 20719 ACK
[Sep  8 17:16:22] Max-Forwards: 70
[Sep  8 17:16:22] Content-Length: 0
[Sep  8 17:16:22]
[Sep  8 17:16:22] <------------->
[Sep  8 17:16:22] --- (8 headers 0 lines) ---
localhost*CLI>

My guess is that you have a broken DNS configuration that is timing out rather than giving either an answer or a definitive not found… Otherwise, Asterisk is waiting for a lock and you would need to know what else was going on at the time.

hi
my dns was indeed broken
it was point to 127.0.0.1 and then to 8.8.8.8
I canged it to 8.8.8.8 8.8.4.4 and restart the machine, but no help.
why the dns is required at that point? my trunk written as ip and not as host name.

Thanks

you gave me good direction.
it was solved by remove the stun server

that was
stun.l.google.com:19302
any idea why?
can’t i use stun server?