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Asterisk和Twilio的弹性SIP中继（入站故障排除）

分类: 技术问答 • 2022-08-01 22:54:18
问题描述：

我已经在这几天，似乎无法将来电路由到用户扩展。然而，当拨打与Twilio Elastic SIP中继相关联的号码时，拨出呼叫和内部SIP分机拨号都可以工作。我已经为域设置和配置了电话，我的电信公司收到了“所有电路都很忙”的消息。Asterisk和Twilio的弹性SIP中继（入站故障排除）

该系统是在Ubuntu 14.04上运行的FreePBX 12.0.68全新安装，内部SIP扩展拨号和外线呼叫在中继线上工作。为Asterisk的Twilio中继配置从here采取和here

type=peer 
secret=xxxxxxxxxxxxxxxxxxx 
username=xxxxxxxxxxxxxxx 
host=xxxxxxxxx.pstn.twilio.com 
dtmfmode=rfc2833 
canreinvite=no 
disallow=all 
allow=ulaw 
insecure=port,invite 
fromuser=xxxxxxxxxxx 
fromdomain=xxxxxxxxx.pstn.twilio.com 
context=incoming

这里的Twilio之间的TCP/UDP流量的服务器

Source   Destination  Protocal Info 
10x.xxx.xx.xxx 10x.xxx.xxx.xx UDP  Source port: 5060 Destination port: 5060 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.2  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.3  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.0  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.0  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
54.172.60.0  10x.xxx.xxx.xx SIP/SDP Request: INVITE sip:[email protected] 
10x.xxx.xx.xxx 10x.xxx.xxx.xx UDP  Source port: 5060 Destination port: 5060

而这里的INVITE UDP流

INVITE sip:[email protected] SIP/2.0 
Record-Route: <sip:54.172.60.0:5060;lr;ftag=11540065_6772d868_144031e0-db91-45e9-ae85-6de18ed14b19> 
From: <sip:[email protected];pstn-params=808481808882;cpc=ordinary>;tag=11540065_6772d868_144031e0-db91-45e9-ae85-6de18ed14b19 
To: <sip:[email protected].com;user=phone> 
CSeq: 25149 INVITE 
Max-Forwards: 132 
Accept: application/sdp,application/isup,application/dtmf,application/dtmf-relay,multipart/mixed 
Session-Expires: 1800 
Min-SE: 90 
Content-Disposition: session;handling=required 
Diversion: sip:[email protected];reason=unconditional 
Call-ID: [email protected] 
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bKdf6c.854803a7.0 
Via: SIP/2.0/UDP 172.18.18.39:5060;branch=z9hG4bK144031e0-db91-45e9-ae85-6de18ed14b19_6772d868_287964010429808 
Contact: <sip:[email protected]:5060;transport=udp> 
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE 
User-Agent: Twilio Gateway 
X-Twilio-AccountSid: ACaa6e5a9a0d40b2b12751f33b612ebf6e 
X-Twilio-ApiVersion: 2010-04-01 
Content-Type: application/sdp 
X-Twilio-CallSid: CAcc7d0e0603fea476fdaa1c94d9243104 
Content-Length: 233 

v=0 
o=- 412164138 412164138 IN IP4 54.172.60.23 
s=SIP Media Capabilities 
c=IN IP4 54.172.60.23 
t=0 0 
m=audio 11590 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=sendrecv 
a=ptime:20 
INVITE sip:[email protected] SIP/2.0 
Record-Route: <sip:54.172.60.0:5060;lr;ftag=11540065_6772d868_144031e0-db91-45e9-ae85-6de18ed14b19> 
From: <sip:[email protected];pstn-params=808481808882;cpc=ordinary>;tag=11540065_6772d868_144031e0-db91-45e9-ae85-6de18ed14b19 
To: <sip:[email protected];user=phone> 
CSeq: 25149 INVITE 
Max-Forwards: 132 
Accept: application/sdp,application/isup,application/dtmf,application/dtmf-relay,multipart/mixed 
Session-Expires: 1800 
Min-SE: 90 
Content-Disposition: session;handling=required 
Diversion: sip:[email protected];reason=unconditional 
Call-ID: [email protected] 
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bKdf6c.854803a7.0 
Via: SIP/2.0/UDP 172.18.18.39:5060;branch=z9hG4bK144031e0-db91-45e9-ae85-6de18ed14b19_6772d868_287964010429808 
Contact: <sip:[email protected]:5060;transport=udp> 
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE 
User-Agent: Twilio Gateway 
X-Twilio-AccountSid: ACaa6e5a9a0d40b2b12751f33b612ebf6e 
X-Twilio-ApiVersion: 2010-04-01 
Content-Type: application/sdp 
X-Twilio-CallSid: CAcc7d0e0603fea476fdaa1c94d9243104 
Content-Length: 233 

v=0 
o=- 412164138 412164138 IN IP4 54.172.60.23 
s=SIP Media Capabilities 
c=IN IP4 54.172.60.23 
t=0 0 
m=audio 11590 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=sendrecv 
a=ptime:20 
INVITE sip:[email protected] SIP/2.0 
Record-Route: <sip:54.172.60.0:5060;lr;ftag=11540065_6772d868_144031e0-db91-45e9-ae85-6de18ed14b19> 
From: <sip:[email protected];pstn-params=808481808882;cpc=ordinary>;tag=11540065_6772d868_144031e0-db91-45e9-ae85-6de18ed14b19 
To: <sip:[email protected];user=phone> 
CSeq: 25149 INVITE 
Max-Forwards: 132 
Accept: application/sdp,application/isup,application/dtmf,application/dtmf-relay,multipart/mixed 
Session-Expires: 1800 
Min-SE: 90 
Content-Disposition: session;handling=required 
Diversion: sip:[email protected];reason=unconditional 
Call-ID: [email protected] 
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bKdf6c.854803a7.0 
Via: SIP/2.0/UDP 172.18.18.39:5060;branch=z9hG4bK144031e0-db91-45e9-ae85-6de18ed14b19_6772d868_287964010429808 
Contact: <sip:[email protected]:5060;transport=udp> 
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE 
User-Agent: Twilio Gateway 
X-Twilio-AccountSid: ACaa6e5a9a0d40b2b12751f33b612ebf6e 
X-Twilio-ApiVersion: 2010-04-01 
Content-Type: application/sdp 
X-Twilio-CallSid: CAcc7d0e0603fea476fdaa1c94d9243104 
Content-Length: 233 

v=0 
o=- 412164138 412164138 IN IP4 54.172.60.23 
s=SIP Media Capabilities 
c=IN IP4 54.172.60.23 
t=0 0 
m=audio 11590 RTP/AVP 0 101 
a=rtpmap:0 PCMU/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=sendrecv 
a=ptime:20

我还有一个在FreePBX接口中配置的传入路由，其中​​DID Number设置为我的Twilio号码，Destination直接设置为用户的SIP分机，并且相应的客户机正在运行并准备好接收呼叫。我已经使用了netstat和tcpdump这对我来说看起来像是从Twilio发送的INVITE请求，而FreePBX只是没有正确路由它？

+0

嘿，我是Twilio的开发者传道人。我个人的SIP知识不会帮助你。我只想说，如果你在这里找不到答案，那么请发邮件给我们的帮助团队[email protected]，他们会找到能够帮助你的人。 – philnash

+0

事实证明，这是动态IP v.s的问题。适用于Twilio的NAT设置 - http://community.freepbx.org/t/asterisk-and-twilios-elastic-sip-trunking-inbound-troubleshooting/29827 – dcd018

+0

嗯，好的。很高兴听到你把它分类！ – philnash

答

我读过你有一个NAT问题 - 我仍然想发布这个人谁也遇到过类似的问题。

我已经从我的几个PBX系统以及我们的一些客户的ViciDIAL开源拨号配置中获得了Twilio Elastic SIP干线的正常工作。我有一段时间似乎与OP描述的内容完全相同。

当然，我只测试了PBX上的入站 - 我尝试了几种不同的配置排列，直到我确定了使用Twilio与世界上任何其他SIP提供商相对的事情。

首先，我有星号的设置的唯一的事：

[twilio] 
host=xxxxxxx.pstn.twilio.com 
type=friend 
dtmfmode=rfc4733 
canreinivite=no 
insecure=port,invite 

[twilio1] 
type=friend 
insecure=port,invite 
host=54.172.60.0 
dtmfmode=rfc4733 
canreinivite=no 

[twilio2] 
type=friend 
insecure=port,invite 
host=54.172.60.1 
dtmfmode=rfc4733 
canreinivite=no

...等你们每个人将使用入站的Twilio IP的，因为你永远不知道这附近的它会利用地理IP的。 （我会注意到星号日志显示通过取决于您的地理区域的知识产权来循环访问）。

你可以在这里看到我的星号的日志，在那里我只带（twilio_phone）取代我twilio电话号码为例隐私：http://pastebin.com/rXz7cY39

的另一个主要区别：

呼入路由（DID的）必须与领先的+号声明，这是不是在我的经验典型

所以，用twilio你必须包括每一个可能的IP设置为Trunk，也号码前包括+两个拨号时out和crea你的入境路线

希望这是一个帮助任何人努力实现！

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