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尝试和确认呼叫转移时忽略的udp SIP数据包（pjsip）

udp asterisk sip

尝试和确认呼叫转移时忽略的udp SIP数据包（pjsip）,udp,asterisk,sip,pjsip,Udp,Asterisk,Sip,Pjsip,我想将调用从一个AGI控制的上下文转移到另一个端点 就是。我有一个cisco网关发起的调用给我的asterisk配置，该配置使用AGI处理一些树逻辑，并最终将调用转移到另一个端点（vbox） 问题是何时发生转移。不知何故，来自vbox的TRYING和OK SIP数据包被星号忽略，因为它们没有显示在CLI中，但通过udp tcpdump可以清楚地接收到。最后，在我的星号重新发出invite数据包之后，调用被丢弃，就好像没有收到trying/ok一样，而实际上它是在接口级别发送和接收的 星号版本15

我想将调用从一个AGI控制的上下文转移到另一个端点

就是。我有一个cisco网关发起的调用给我的asterisk配置，该配置使用AGI处理一些树逻辑，并最终将调用转移到另一个端点（vbox）

问题是何时发生转移。不知何故，来自vbox的TRYING和OK SIP数据包被星号忽略，因为它们没有显示在CLI中，但通过udp tcpdump可以清楚地接收到。最后，在我的星号重新发出invite数据包之后，调用被丢弃，就好像没有收到trying/ok一样，而实际上它是在接口级别发送和接收的

星号版本15.5.0

pjsip.conf

[t-udp-m]
type=transport
protocol=udp
bind=62.12.240.121:5060

[gw1]
type=endpoint
transport=t-udp-m
context=civr
disallow=all
allow=ulaw,speex,gsm
aors=gw1

[gw1]
type=identify
endpoint=gw1
match=62.15.164.62

[gw1]
type=aor
max_contacts=2
remove_existing=yes

[vbox]
type=endpoint
transport=t-udp-m
context=civr
disallow=all
allow=ulaw,alaw,gsm,speex
aors=vbox
send_rpid=yes

[vbox]
type=identify
endpoint=vbox
match=62.15.164.65

[vbox]
type=aor
contact=sip:7011@62.15.164.65:5060
max_contacts=1
remove_existing=yes
extensions.conf

[civr]
exten => 7010,1,Answer()
 same => n,Agi(/var/www/agi-bin/agi.php)
 same => n,Hangup()

[call_center_altitude]
exten => 1,1,Dial(PJSIP/vbox)
 same => n,Hangup()
tcpdump

IP 62.15.164.62.61396 > 62.12.240.121.sip: UDP, length 1159
INVITE sip:7010@62.12.240.121:5060 SIP/2.0
Via: SIP/2.0/UDP  62.15.164.62:5060;branch=z9hG4bK4CB8B17BE
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>
Date: Tue, 23 Oct 2018 02:00:59 GMT
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 1380744107-3583906280-2679701539-3944536960
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:0971110799@62.15.164.62>;party=calling;screen=yes;privacy=off
Timestamp: 1540260059
Contact: <sip:0971110799@62.15.164.62:5060>
Call-Info: <sip:62.15.164.62:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 1066 6200 IN IP4 62.15.164.62
s=SIP Call
c=IN IP4 62.15.164.62
t=0 0
m=audio 16658 RTP/AVP 0 101
c=IN IP4 62.15.164.62
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

IP 62.12.240.121.sip > 62.15.164.62.61396: UDP, length 326
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.15.164.62:5060;rport=61396;received=62.15.164.62;branch=z9hG4bK4CB8B17BE
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>
CSeq: 101 INVITE
Server: Asterisk PBX 15.5.0
Content-Length:  0


IP 62.12.240.121.sip > 62.15.164.62.61396: UDP, length 818
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.15.164.62:5060;rport=61396;received=62.15.164.62;branch=z9hG4bK4CB8B17BE
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>;tag=3780211b-39df-482f-8b48-b0ca039a65b2
CSeq: 101 INVITE
Server: Asterisk PBX 15.5.0
Contact: <sip:62.12.240.121:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   227

v=0
o=- 1066 6202 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 26188 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

IP 62.15.164.62.61396 > 62.12.240.121.sip: UDP, length 371
ACK sip:62.12.240.121:5060 SIP/2.0
Via: SIP/2.0/UDP  62.15.164.62:5060;branch=z9hG4bK4CB8CF50
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>;tag=3780211b-39df-482f-8b48-b0ca039a65b2
Date: Tue, 23 Oct 2018 02:00:59 GMT
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

!!!! The problem starts here.....

IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
INVITE sip:7011@62.15.164.65:5060 SIP/2.0
Via: SIP/2.0/UDP 62.12.240.121:5060;rport;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Contact: <sip:asterisk@62.12.240.121:5060>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: <sip:00008577751000971110799@62.12.240.121>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 15.5.0
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 272899537 272899537 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 21388 RTP/AVP 0 8 3 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 564
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.12.240.121:5060;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89;received=62.12.240.121;rport=58140
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Server: Altitude vBox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7011@62.15.164.65:5060>
Content-Length: 0


IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 859
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.12.240.121:5060;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89;received=62.12.240.121;rport=58140
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>;tag=as1dce0520
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Server: Altitude vBox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7011@62.15.164.65:5060>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 140358089 140358089 IN IP4 62.15.164.65
s=Altitude vBox
c=IN IP4 62.15.164.65
t=0 0
m=audio 25364 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

!!!! As you can see, the last 2 packets were ignored by my asterisk configuration and so it re-issued the INVITE packets...

IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
INVITE sip:7011@62.15.164.65:5060 SIP/2.0
Via: SIP/2.0/UDP 62.12.240.121:5060;rport;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Contact: <sip:asterisk@62.12.240.121:5060>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: <sip:00008577751000971110799@62.12.240.121>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 15.5.0
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 272899537 272899537 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 21388 RTP/AVP 0 8 3 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

ip62.15.164.62.61396>62.12.240.121.sip:UDP，长度1159
邀请sip：7010@62.12.240.121：5060 SIP/2.0
Via:SIP/2.0/UDP 62.15.164.62:5060；分支=z9hG4bK4CB8B17BE
发件人：；标签=BD15CD5C-FC6
致：
日期：2018年10月23日星期二02:00:59 GMT
电话号码：524DB3B-D59E11E8-AAA1E790-46F000BF@62.15.164.62
支持：100rel，定时器，替换
Min SE:1800
思科Guid:1380744107-3583906280-2679701539-3944536960
用户代理：Cisco SIPGateway/IOS-12.x
允许：邀请、选项、再见、取消、确认、PRACK、COMET、参考、订阅、通知、信息、更新、注册
CSeq:101邀请
最大前锋：70
远程方ID:；一方=呼叫；屏幕=是；隐私=关闭
时间戳：1540260059
联系人：
通话信息：；方法=“通知；事件=电话事件；持续时间=2000”
有效期：180
允许事件：电话事件
内容类型：应用程序/sdp
内容长度：247
v=0
o=IP4 62.15.164.62中的CiscoSystemsSIP GW用户代理1066 6200
s=SIP呼叫
c=在IP4 62.15.164.62中
t=0
m=音频16658 RTP/AVP 0 101
c=在IP4 62.15.164.62中
a=rtpmap:0 PCMU/8000
a=rtpmap:101电话事件/8000
a=fmtp:101 0-16
a=时间：20
IP 62.12.240.121.sip>62.15.164.62.61396:UDP，长度326
SIP/2.0 100
Via:SIP/2.0/UDP 62.15.164.62:5060；rport=61396；收到=62.15.164.62；分支=z9hG4bK4CB8B17BE
电话号码：524DB3B-D59E11E8-AAA1E790-46F000BF@62.15.164.62
发件人：；标签=BD15CD5C-FC6
致：
CSeq:101邀请
服务器：星号PBX 15.5.0
内容长度：0
IP 62.12.240.121.sip>62.15.164.62.61396:UDP，长度818
SIP/2.0 200正常
Via:SIP/2.0/UDP 62.15.164.62:5060；rport=61396；收到=62.15.164.62；分支=z9hG4bK4CB8B17BE
电话号码：524DB3B-D59E11E8-AAA1E790-46F000BF@62.15.164.62
发件人：；标签=BD15CD5C-FC6
致：；标签=3780211b-39df-482f-8b48-b0ca039a65b2
CSeq:101邀请
服务器：星号PBX 15.5.0
联系人：
允许：选项、订阅、通知、发布、邀请、确认、再见、取消、更新、恶作剧、注册、消息、引用
支持：100rel、计时器、替换、norefersub
内容类型：应用程序/sdp
内容长度：227
v=0
o=-1066 6202在IP4 62.12.240.121中
s=星号
c=在IP4 62.12.240.121中
t=0
m=音频26188 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101电话事件/8000
a=fmtp:101 0-16
a=时间：20
a=最大时间：150
a=sendrecv
IP 62.15.164.62.61396>62.12.240.121.sip:UDP，长度371
确认sip:62.12.240.121:5060 sip/2.0
Via:SIP/2.0/UDP 62.15.164.62:5060；分支=z9hG4bK4CB8CF50
发件人：；标签=BD15CD5C-FC6
致：；标签=3780211b-39df-482f-8b48-b0ca039a65b2
日期：2018年10月23日星期二02:00:59 GMT
电话号码：524DB3B-D59E11E8-AAA1E790-46F000BF@62.15.164.62
最大前锋：70
CSeq:101确认
内容长度：0
!!!! 问题从这里开始。。。。。
IP 62.12.240.121.sip>62.15.164.65.sip:UDP，长度1068
邀请sip：7011@62.15.164.65：5060 SIP/2.0
Via:SIP/2.0/UDP 62.12.240.121:5060；rport；分支=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
发件人：；标签=b2e6cd02-127c-42ca-a51a-f98cb26329aa
致：
联系人：
呼叫ID:6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq:11604邀请
允许：选项、订阅、通知、发布、邀请、确认、再见、取消、更新、恶作剧、注册、消息、引用
支持：100rel、计时器、替换、norefersub
会期届满：1800
Min SE:90
远程方ID:；隐私=关闭；屏幕=否
最大前锋：70
用户代理：星号PBX 15.5.0
内容类型：应用程序/sdp
内容长度：312
v=0
o=-272899537 IP4 62.12.240.121中的272899537
s=星号
c=在IP4 62.12.240.121中
t=0
m=音频21388 RTP/AVP 0 8 3 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101电话事件/8000
a=fmtp:101 0-16
a=时间：20
a=最大时间：60
a=sendrecv
IP 62.15.164.65.sip>62.12.240.121.58140:UDP，长度564
SIP/2.0 100
Via:SIP/2.0/UDP 62.12.240.121:5060；分支=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89；已接收=62.12.240.121；rport=58140
发件人：；标签=b2e6cd02-127c-42ca-a51a-f98cb26329aa
致：
呼叫ID:6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq:11604邀请
服务器：高度vBox
允许：邀请、确认、取消、选项、再见、参考、订阅、通知、信息、发布
支持：替换、定时器
届会结束：1800；复习者=无人机
联系人：
内容长度：0
IP 62.15.164.65.sip>62.12.240.121.58140:UDP，长度859
SIP/2.0 200正常
Via:SIP/2.0/UDP 62.12.240.121:5060；分支=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89；已接收=62.12.240.121；rport=58140
发件人：；标签=b2e6cd02-127c-42ca-a51a-f98cb26329aa
致：；标签=as1dce0520
呼叫ID:6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq:11604邀请
服务器：高度vBox
允许：邀请、确认、取消、选项、再见、参考、订阅、通知、信息、发布
支持：替换、定时器
届会结束：1800；复习者=无人机
联系人：
内容类型：应用程序/sdp
内容长度：251
v=0
o=IP4 62.15.164.65中的根140358089 140358089
s=高度vBox
c=在IP4 62.15.164.65中
t=0
m=音频25364 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101电话事件/8000
a=fmtp:101 0-16
a=时间：20
a=sendrecv
!!!! 如您所见，我的星号配置忽略了最后2个数据包，因此它重新发出了INVITE数据包。。。
IP 62.12.240.121.sip>62.15.164.65.sip:UDP，长度1068
邀请sip：7011@62.15.164.65：5060 SIP/2.0
Via:SIP/2.0/UDP 62.12.240.121:5060；rport；分支=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
发件人：；标签=b2e6cd02-127c-42ca-a51a-f98cb26329aa
致：
康塔克

IP 62.15.164.62.61396 > 62.12.240.121.sip: UDP, length 1159
INVITE sip:7010@62.12.240.121:5060 SIP/2.0
Via: SIP/2.0/UDP  62.15.164.62:5060;branch=z9hG4bK4CB8B17BE
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>
Date: Tue, 23 Oct 2018 02:00:59 GMT
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 1380744107-3583906280-2679701539-3944536960
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:0971110799@62.15.164.62>;party=calling;screen=yes;privacy=off
Timestamp: 1540260059
Contact: <sip:0971110799@62.15.164.62:5060>
Call-Info: <sip:62.15.164.62:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 1066 6200 IN IP4 62.15.164.62
s=SIP Call
c=IN IP4 62.15.164.62
t=0 0
m=audio 16658 RTP/AVP 0 101
c=IN IP4 62.15.164.62
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

IP 62.12.240.121.sip > 62.15.164.62.61396: UDP, length 326
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.15.164.62:5060;rport=61396;received=62.15.164.62;branch=z9hG4bK4CB8B17BE
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>
CSeq: 101 INVITE
Server: Asterisk PBX 15.5.0
Content-Length:  0


IP 62.12.240.121.sip > 62.15.164.62.61396: UDP, length 818
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.15.164.62:5060;rport=61396;received=62.15.164.62;branch=z9hG4bK4CB8B17BE
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>;tag=3780211b-39df-482f-8b48-b0ca039a65b2
CSeq: 101 INVITE
Server: Asterisk PBX 15.5.0
Contact: <sip:62.12.240.121:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   227

v=0
o=- 1066 6202 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 26188 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

IP 62.15.164.62.61396 > 62.12.240.121.sip: UDP, length 371
ACK sip:62.12.240.121:5060 SIP/2.0
Via: SIP/2.0/UDP  62.15.164.62:5060;branch=z9hG4bK4CB8CF50
From: <sip:0971110799@62.15.164.62>;tag=BD15CD5C-FC6
To: <sip:7010@62.12.240.121>;tag=3780211b-39df-482f-8b48-b0ca039a65b2
Date: Tue, 23 Oct 2018 02:00:59 GMT
Call-ID: 524DB3FB-D59E11E8-AAA1E790-46F000BF@62.15.164.62
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

!!!! The problem starts here.....

IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
INVITE sip:7011@62.15.164.65:5060 SIP/2.0
Via: SIP/2.0/UDP 62.12.240.121:5060;rport;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Contact: <sip:asterisk@62.12.240.121:5060>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: <sip:00008577751000971110799@62.12.240.121>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 15.5.0
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 272899537 272899537 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 21388 RTP/AVP 0 8 3 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 564
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.12.240.121:5060;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89;received=62.12.240.121;rport=58140
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Server: Altitude vBox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7011@62.15.164.65:5060>
Content-Length: 0


IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 859
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.12.240.121:5060;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89;received=62.12.240.121;rport=58140
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>;tag=as1dce0520
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Server: Altitude vBox
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7011@62.15.164.65:5060>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 140358089 140358089 IN IP4 62.15.164.65
s=Altitude vBox
c=IN IP4 62.15.164.65
t=0 0
m=audio 25364 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

!!!! As you can see, the last 2 packets were ignored by my asterisk configuration and so it re-issued the INVITE packets...

IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
INVITE sip:7011@62.15.164.65:5060 SIP/2.0
Via: SIP/2.0/UDP 62.12.240.121:5060;rport;branch=z9hG4bKPj702bedc6-b346-44b2-8cc1-bbdffcb75a89
From: <sip:00008577751000971110799@62.12.240.121>;tag=b2e6cd02-127c-42ca-a51a-f98cb26329aa
To: <sip:7011@62.15.164.65>
Contact: <sip:asterisk@62.12.240.121:5060>
Call-ID: 6de2b47c-8264-452a-9656-d7d4ed6b731f
CSeq: 11604 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: <sip:00008577751000971110799@62.12.240.121>;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 15.5.0
Content-Type: application/sdp
Content-Length:   312

v=0
o=- 272899537 272899537 IN IP4 62.12.240.121
s=Asterisk
c=IN IP4 62.12.240.121
t=0 0
m=audio 21388 RTP/AVP 0 8 3 110 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv



<--- Received SIP request (1159 bytes) from UDP:62.15.164.62:61396 --->
<INVITE data same as in tcpdump>
  == Setting global variable 'SIPDOMAIN' to '62.12.240.121'
<--- Transmitting SIP response (326 bytes) to UDP:62.15.164.62:61396 --->
<TRYING data same as in tcpdump>
    -- Executing [7010@civr:1] Answer("PJSIP/gw1-00000022", "") in new stack
       > 0x7f707c019640 -- Strict RTP learning after remote address set to: 62.15.164.62:16658
<--- Transmitting SIP response (818 bytes) to UDP:62.15.164.62:61396 --->
<OK data same as in tcpdump>
<--- Received SIP request (371 bytes) from UDP:62.15.164.62:61396 --->
<ACK data same as in tcpdump>
       > 0x7f707c019640 -- Strict RTP switching to RTP target address 62.15.164.62:16658 as source
    -- Executing [7010@civr:8] AGI("PJSIP/gw1-00000022", "/var/www/agi-bin/agi.php") in new stack
    -- Launched AGI Script /var/www/agi-bin/agi.php
<PJSIP/gw1-00000022>AGI Tx >> agi_request: /var/www/agi-bin/agi.php
<PJSIP/gw1-00000022>AGI Tx >> agi_channel: PJSIP/gw1-00000022
<PJSIP/gw1-00000022>AGI Tx >> agi_language: en
<PJSIP/gw1-00000022>AGI Tx >> agi_type: PJSIP
<PJSIP/gw1-00000022>AGI Tx >> agi_uniqueid: 1540257639.63
<PJSIP/gw1-00000022>AGI Tx >> agi_version: 15.5.0
<PJSIP/gw1-00000022>AGI Tx >> agi_callerid: 0971110799
<PJSIP/gw1-00000022>AGI Tx >> agi_calleridname: unknown
<PJSIP/gw1-00000022>AGI Tx >> agi_callingpres: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_callingani2: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_callington: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_callingtns: 0
<PJSIP/gw1-00000022>AGI Tx >> agi_dnid: 7010
<PJSIP/gw1-00000022>AGI Tx >> agi_rdnis: unknown
<PJSIP/gw1-00000022>AGI Tx >> agi_context: civr
<PJSIP/gw1-00000022>AGI Tx >> agi_extension: 7010
<PJSIP/gw1-00000022>AGI Tx >> agi_priority: 8
<PJSIP/gw1-00000022>AGI Tx >> agi_enhanced: 0.0
<PJSIP/gw1-00000022>AGI Tx >> agi_accountcode:
<PJSIP/gw1-00000022>AGI Tx >> agi_threadid: 140122966906624
<PJSIP/gw1-00000022>AGI Tx >>
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-yVABhE ""
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-yVABhE.slin' (escape_digits=) (sample_offset 0) (language 'en')
       > 0x7f707c019640 -- Strict RTP learning complete - Locking on source address 62.15.164.62:16658
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0 endpos=47554
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-9pPsxq "1234"
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-9pPsxq.slin' (escape_digits=1234) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=50 endpos=21920
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-zztyOj "12"
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-zztyOj.slin' (escape_digits=12) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=49 endpos=12320
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-FyUq0g "#*0123456789"
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-FyUq0g.slin' (escape_digits=#*0123456789) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=56 endpos=13440
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=53
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=55
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=55
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=55
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=53
<PJSIP/gw1-00000022>AGI Rx << WAIT FOR DIGIT 2000
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << SET CONTEXT call_center_altitude
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << SET CALLERID 00008577751000971110799
<PJSIP/gw1-00000022>AGI Tx >> 200 result=1
<PJSIP/gw1-00000022>AGI Rx << SET EXTENSION 1
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << SET PRIORITY 1
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0
<PJSIP/gw1-00000022>AGI Rx << STREAM FILE /tmp/fest-e3ELZu ""
    -- <PJSIP/gw1-00000022> Playing '/tmp/fest-e3ELZu.slin' (escape_digits=) (sample_offset 0) (language 'en')
<PJSIP/gw1-00000022>AGI Tx >> 200 result=0 endpos=34697
    -- <PJSIP/gw1-00000022>AGI Script /var/www/agi-bin/agi.php completed, returning 0
    -- Executing [1@call_center_altitude:1] Dial("PJSIP/gw1-00000022", "PJSIP/vbox") in new stack
    -- Called PJSIP/vbox
<--- Transmitting SIP request (1068 bytes) to UDP:62.15.164.65:5060 --->
<INVITE same as in tcpdump>



IP 62.12.240.121.sip > 62.15.164.65.sip: UDP, length 1068
IP 62.15.164.65.sip > 62.12.240.121.58140: UDP, length 564