O asterisco expira e finaliza a conexão após 6400ms devido a 'sem resposta'

Question

Eu tenho um tronco SIP configurado com o Twilio para chamadas de saída. O Twilio-FreePBX e o meu dispositivo de teste é o X-Lite simples do CounterPath.

Eu posso fazer uma chamada de saída do X-Lite. Meu celular toca e eu posso atender. Mas é isso, não há áudio transmitido e a chamada desliga sozinha após alguns segundos.

Este é o erro que recebo do log do Asterisk dentro do meu servidor FreePBX:

[2017-02-17 14:41:46] WARNING[1996] chan_sip.c: Retransmission timeout reached     on transmission 83369MWU3MmY5MWZiZWZkODJmYjc3ZWEzMWI5ZmQzMTQ1NWQ for seqno 2   (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2017-02-17 14:41:46] WARNING[1996] chan_sip.c: Hanging up call 83369MWU3MmY5MWZiZWZkODJmYjc3ZWEzMWI5ZmQzMTQ1NWQ - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

A chamada é encaminhada através da minha conta Twilio, posso ver nos logs lá. Registra como complete .

Liguei o firewall FreePBX e adicionei IPs confiáveis

OlogdedepuraçãocompletodoAsterisk:

<------------->[2017-02-1715:18:58]VERBOSE[1996]chan_sip.c:---(12headers12lines)---[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:FoundRTPaudioformat0[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:FoundRTPaudioformat101[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:FoundaudiodescriptionformatPCMUforID0[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:Foundaudiodescriptionformattelephone-eventforID101[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:Capabilities:us-(ulaw|alaw|gsm|g726),peer-audio=(ulaw)/video=(nothing)/text=(nothing),combined-(ulaw)[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:Non-codeccapabilities(dtmf):us-0x1(telephone-event|),peer-0x1(telephone-event|),combined-0x1(telephone-event|)[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:PeeraudioRTPisatport54.172.61.111:12510[2017-02-1715:18:58]VERBOSE[1996][C-00000004]sip/route.c:sip_route_dump:route/pathhop:<sip:54.172.60.2:5060;lr;ftag=as7484893d;twnat=sip:52.90.124.243:5060>[2017-02-1715:18:58]VERBOSE[1996][C-00000004]chan_sip.c:Transmitting(NAT)to54.172.60.2:5060:ACKsip:172.18.7.119:5060SIP/2.0Via:SIP/2.0/UDP172.**.**.***:5060;branch=z9hG4bK2b6c05a0;rportRoute:<sip:54.172.60.2:5060;lr;ftag=as7484893d;twnat=sip:52.90.124.243:5060>Max-Forwards:70From:<sip:PHIL@172.**.**.***>;tag=as7484893dTo:<sip:[phone]@********.pstn.twilio.com>;tag=77864250_6772d868_655d5c53-0b14-4aa5-8bd5-d8f83501d26cContact:<sip:PHIL@172.**.**.***:5060>Call-ID:664a272c08f7af0543b2bac950391d32@172.**.**.***:5060CSeq:103ACKUser-Agent:FPBX-13.0.190.7(13.12.2)Content-Length:0---[2017-02-1715:18:58]VERBOSE[8613][C-00000004]app_dial.c:SIP/TwilioTrunk-00000009answeredSIP/808-00000008[2017-02-1715:18:58]VERBOSE[8613][C-00000004]chan_sip.c:Audioisat12824[2017-02-1715:18:58]VERBOSE[8613][C-00000004]chan_sip.c:AddingcodeculawtoSDP[2017-02-1715:18:58]VERBOSE[8613][C-00000004]chan_sip.c:AddingcodecalawtoSDP[2017-02-1715:18:58]VERBOSE[8613][C-00000004]chan_sip.c:Addingnon-codec0x1(telephone-event)toSDP[2017-02-1715:18:58]VERBOSE[8613][C-00000004]chan_sip.c:<---ReliablyTransmitting(NAT)to73.81.116.96:35304--->SIP/2.0200OKVia:SIP/2.0/UDP73.81.116.96:35304;branch=z9hG4bK-524287-1---d5cf0232413c0269;received=73.81.116.96;rport=35304From:"Phil"<sip:[email protected]>;tag=82678409
To: <sip:[email protected]>;tag=as08c320c9
Call-ID: 83369ZWQ2OTI2MjRkNGE3MTdlYmM5MjYxM2Q0ZDIwOWVhYTM
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[phone]@172.**.**.***:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1504626483 1504626483 IN IP4 172.**.**.***
s=Asterisk PBX 13.12.2
c=IN IP4 172.**.**.***
t=0 0
m=audio 12824 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
[2017-02-17 15:18:58] VERBOSE[8637][C-00000004] bridge_channel.c: Channel SIP/Twilio Trunk-00000009 joined 'simple_bridge' basic-bridge <e035a287-8fa3-4291-a3ad-927bca1407e0>
[2017-02-17 15:18:58] VERBOSE[8613][C-00000004] bridge_channel.c: Channel SIP/808-00000008 joined 'simple_bridge' basic-bridge <e035a287-8fa3-4291-a3ad-927bca1407e0>
[2017-02-17 15:18:58] VERBOSE[1996] chan_sip.c: Retransmitting #1 (NAT) to 73.81.116.96:35304:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 73.81.116.96:35304;branch=z9hG4bK-524287-1---d5cf0232413c0269;received=73.81.116.96;rport=35304
From: "Phil"<sip:[email protected]>;tag=82678409
To: <sip:[email protected]>;tag=as08c320c9
Call-ID: 83369ZWQ2OTI2MjRkNGE3MTdlYmM5MjYxM2Q0ZDIwOWVhYTM
CSeq: 2 INVITE
Server: FPBX-13.0.190.7(13.12.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[phone]@172.**.**.***:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1504626483 1504626483 IN IP4 172.**.**.***
s=Asterisk PBX 13.12.2
c=IN IP4 172.**.**.***
t=0 0
m=audio 12824 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

O Retransmitting continua por 6 tentativas antes de eliminar a conexão.

Qualquer ajuda seria ótima. Obrigado antecipadamente!