星号超时并在6400ms后终止连接,由于“无响应”

我有一个SIP中继与Twilio设置为出站呼叫。 Twilio-FreePBX,然后我的testing设备是CounterPath的简单的X-Lite。

我可以从X-Lite拨打电话。 我的手机响了,我可以拿起。 但就是这样,没有audio传​​输,并在几秒钟后自行挂断电话。

这是我从我的FreePBX服务器中的Asterisk日志收到的错误:

[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). 

这个电话是通过我的Twilio帐户路由,我可以在那里看到它的日志。 它注册为complete

我已经打开了FreePBX防火墙并添加了可信的IP 在这里输入图像说明

完整的Asteriskdebugging日志:

 <-------------> [2017-02-17 15:18:58] VERBOSE[1996] chan_sip.c: --- (12 headers 12 lines) --- [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found RTP audio format 0 [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found RTP audio format 101 [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found audio description format PCMU for ID 0 [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Found audio description format telephone-event for ID 101 [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Peer audio RTP is at port 54.172.61.111:12510 [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] sip/route.c: sip_route_dump: route/path hop: <sip:54.172.60.2:5060;lr;ftag=as7484893d;twnat=sip:52.90.124.243:5060> [2017-02-17 15:18:58] VERBOSE[1996][C-00000004] chan_sip.c: Transmitting (NAT) to 54.172.60.2:5060: ACK sip:172.18.7.119:5060 SIP/2.0 Via: SIP/2.0/UDP 172.**.**.***:5060;branch=z9hG4bK2b6c05a0;rport Route: <sip:54.172.60.2:5060;lr;ftag=as7484893d;twnat=sip:52.90.124.243:5060> Max-Forwards: 70 From: <sip:PHIL@172.**.**.***>;tag=as7484893d To: <sip:+18566492240@********.pstn.twilio.com>;tag=77864250_6772d868_655d5c53-0b14-4aa5-8bd5-d8f83501d26c Contact: <sip:PHIL@172.**.**.***:5060> Call-ID: 664a272c08f7af0543b2bac950391d32@172.**.**.***:5060 CSeq: 103 ACK User-Agent: FPBX-13.0.190.7(13.12.2) Content-Length: 0 --- [2017-02-17 15:18:58] VERBOSE[8613][C-00000004] app_dial.c: SIP/Twilio Trunk-00000009 answered SIP/808-00000008 [2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Audio is at 12824 [2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Adding codec ulaw to SDP [2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Adding codec alaw to SDP [2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2017-02-17 15:18:58] VERBOSE[8613][C-00000004] chan_sip.c: <--- Reliably Transmitting (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:+18566492240@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:+18566492240@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 

在重新连接之前, Retransmitting进行了6次尝试。

任何帮助将是伟大的。 提前致谢!