在更改提供者的Cisco模式后，星号不携带来电。
我的星号（“ my_ip”）回答了BYE的到来。
有人可以问我为什么吗？ （收集的数据）
<--- SIP read from UDP:<<incoming_ip>>:5060 --->
INVITE sip:<<our_in_phone>>@B2B-EKB-Sung:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP <<incoming_ip>>:5060;branch=z9hG4bKnndg8g009o0b66poa600.1
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS,REGISTER,REFER,SUBSCRIBE,MESSAGE,PUBLISH
Call-ID: 461f392oriy4qgcdk@172.16.5.3
Contact: "<<incomin_mobile>>" <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;transport=udp>
CSeq: 250 INVITE
Expires: 3600
From: "<<incomin_mobile>>" <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;cpc=ordinary>;tag=hgdjlmovjl
To: "<<our_in_phone>>" <sip:<<our_in_phone>>@<<our_ip>>:5060;user=phone>
Max-Forwards: 69
Privacy: none
Content-Length: 243
Content-Type: application/sdp
v=0
o=- 1212881 2288486 IN IP4 <<incoming_ip>>
s=-
c=IN IP4 <<incoming_ip>>
b=AS:64
t=0 0
m=audio 19062 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Sending to <<incoming_ip>>:5060 (no NAT)
Using INVITE request as basis request - 461f392oriy4qgcdk@172.16.5.3
Found peer 'utk' for '<<incomin_mobile>>' from <<incoming_ip>>:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <<incoming_ip>>:19062
Looking for <<our_in_phone>> in IN (domain B2B-EKB-Sung)
list_route: hop: <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;transport=udp>
<--- Transmitting (no NAT) to <<incoming_ip>>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <<incoming_ip>>:5060;branch=z9hG4bKnndg8g009o0b66poa600.1;received=<<incoming_ip>>
From: "<<incomin_mobile>>" <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;cpc=ordinary>;tag=hgdjlmovjl
To: "<<our_in_phone>>" <sip:<<our_in_phone>>@<<our_ip>>:5060;user=phone>
Call-ID: 461f392oriy4qgcdk@172.16.5.3
CSeq: 250 INVITE
Server: sunshamil@yandex.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<<our_in_phone>>@<<our_ip>>:5060>
Content-Length: 0
<------------>
Audio is at 10050
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to <<incoming_ip>>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <<incoming_ip>>:5060;branch=z9hG4bKnndg8g009o0b66poa600.1;received=<<incoming_ip>>
From: "<<incomin_mobile>>" <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;cpc=ordinary>;tag=hgdjlmovjl
To: "<<our_in_phone>>" <sip:<<our_in_phone>>@<<our_ip>>:5060;user=phone>;tag=as3662657b
Call-ID: 461f392oriy4qgcdk@172.16.5.3
CSeq: 250 INVITE
Server: sunshamil@yandex.ru
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:<<our_in_phone>>@<<our_ip>>:5060>
Content-Type: application/sdp
Content-Length: 235
v=0
o=root 2100145274 2100145274 IN IP4 <<our_ip>>
s=Asterisk PBX 11.2.1
c=IN IP4 <<our_ip>>
t=0 0
m=audio 10050 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
白包最后没有“好”的内容。也许是合并的数据包
<--- SIP read from UDP:<<incoming_ip>>:5060 --->
ACK sip:<<our_in_phone>>@<<our_ip>>:5060 SIP/2.0
Via: SIP/2.0/UDP <<incoming_ip>>:5060;branch=z9hG4bKsf83b130a8q0c35do500.1
Call-ID: 461f392oriy4qgcdk@172.16.5.3
CSeq: 250 ACK
From: "<<incomin_mobile>>" <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;cpc=ordinary>;tag=hgdjlmovjl
To: "<<our_in_phone>>" <sip:<<our_in_phone>>@<<our_ip>>:5060;user=phone>;tag=as3662657b
Max-Forwards: 69
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '461f392oriy4qgcdk@172.16.5.3' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;transport=udp> for address/port to send to
set_destination: set destination to <<incoming_ip>>:5060
Reliably Transmitting (no NAT) to <<incoming_ip>>:5060:
BYE sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP <<our_ip>>:5060;branch=z9hG4bK1853b292
Max-Forwards: 70
From: "<<our_in_phone>>" <sip:<<our_in_phone>>@<<our_ip>>:5060;user=phone>;tag=as3662657b
To: "<<incomin_mobile>>" <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;cpc=ordinary>;tag=hgdjlmovjl
Call-ID: 461f392oriy4qgcdk@172.16.5.3
CSeq: 102 BYE
User-Agent: sunshamil@yandex.ru
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
如果我理解这是正确的，那么我们的星号会为收到的邮件回答“再见”：
<--- SIP read from UDP:<<incoming_ip>>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <<our_ip>>:5060;branch=z9hG4bK1853b292
From: "<<our_in_phone>>" <sip:<<our_in_phone>>@<<our_ip>>:5060;user=phone>;tag=as3662657b
To: "<<incomin_mobile>>" <sip:<<incomin_mobile>>@<<incoming_ip>>:5060;user=phone;cpc=ordinary>;tag=hgdjlmovjl
Call-ID: 461f392oriy4qgcdk@172.16.5.3
CSeq: 102 BYE
Content-Length: 0
答案 0 :(得分：-1)
解决
这是我的错误，'bye'是来自agi脚本，因为新的cisco不在允许的列表中。