I want to transfer the call from one AGI controlled context to another endpoint.
That is. I have a cisco gateway initiated call given to my asterisk configuration which handles some tree logic with AGI and eventually transfers the call to another endpoint (vbox)
cisco(62.15.164.62)->
asterisk(62.12.240.121)-> do some AGI then transfer to vbox
vbox(62.15.164.65)
The problem is when the transfer happens. Somehow the TRYING and OK SIP packets from vbox are ignored by asterisk as they are not shown in the CLI but are clearly received watching the udp tcpdump. And eventually after my asterisk re-issues the invite packets the call gets dropped as if no trying/ok was received when in fact it was sent and received at the interface level.
Asterisk version 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
Which eventually led to a dropped call.
CLI Output
<--- 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>
!!! TRYING and OK SIP packets does not show in the CLI as shown in the tcpdump!!!
There may be something in my asterisk vbox endpoint configuration that I am having wrong.
tcpdump command is tcpdump -nqt -s 0 -A -i eth0 port 5060 where eth0 is the asterisk not receiving the trying/ack on the CLI 62.12.240.121
My... I'll pay you if you can help me here.
