asterisk early dial(overlap)

sip.conf文件中设置参数 allowoverlap=yes ，没有此参数默认也是yes。 这样asterisk便支持early dial了。

话机拨打若干个号码，如拨打一个号码5，发送SIP INVITE包到asterisk，asterisk收到包后， 进行分析。

如下

<------------->

--- (17 headers 16 lines) ---

Sending to  192.168.122.155:5060 (no NAT)

Using INVITE request as basis request - 2069677588-5060-469@BJC.BGI. BCC.BFF

Found peer '3805' for '3805' from  192.168.122.155:5060

  == Using SIP RTP CoS mark 5

Found RTP audio format 0

Found RTP audio format 101

Found audio description format PCMU for ID 0

Found audio description format telephone-event for ID 101

Found RTP video format 99

Found video description format H264 for ID 99

Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x4 (ulaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

Peer audio RTP is at port  192.168.122.155:55450

Looking for 5 in from-internal (domain 192.168.122.38)

<--- Reliably Transmitting (no NAT) to  192.168.122.155:5060 --->

SIP/2.0 484 Address Incomplete

Via: SIP/2.0/UDP 192.168.122.155:5060;branch= z9hG4bK755996444;received=192. 168.122.155;rport=5060

From: "3805" < sip:3805@192.168.122.38>;tag= 1452114578

To: < sip:5@192.168.122.38>;tag= as0328ad0b

Call-ID: 2069677588-5060-469@BJC.BGI. BCC.BFF

CSeq: 4681 INVITE

Server: Asterisk PBX 1.8.7.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>

asterisk在指定的context中进行遍历查询是否存在 潜在的匹配规则（没有限制号码的长度）， 比如这里的context是freepbx定义的from- internal，这个context可以任意定义。如果存在， 发送484 给源地址，等待源继续发送号码，直到匹配到唯一的拨号规则后， 进入到拨号规则中去一一执行；否认发送404，not found。