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0 / Asterisk Integration
White Paper
Version 0.2
Version 0.2
Version 0.2
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REVISION HISTORY
Revisi
on
Date
Published
Author
Comment
0.1
0.2
Philippe Rais
Philippe Rais
Initial draft
Added note about secret option (paragraph
3.3.1).
Version 0.2
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TABLE OF CONTENTS
TABLE OF CONTENTS
1 ARCHITECTURE
1.1 OVERVIEW
2.1 SUBSCRIPTION
14
3.1 ENVIRONMENT
14
3.2 GENESYS
14
16
3.3.1 sip.conf..................................................................................................................17
3.3.2 extensions.conf.....................................................................................................17
4 ANNEX A PRESENCE SUBSCRIPTION
19
27
29
41
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1 ARCHITECTURE
1.1 OVERVIEW
Stream
Manager
Network
SIP
Asterisk
SIP
SIP Server
TLIB
Genesys
Suite
SIP
TLIB
Endpoints are registered on Asterisk only. They are not registered on SIP
Server.
Agent desktop is required in order to maintain agent status (logged in,
logged out, ready, not ready) toward SIP Server.
Agent desktop is also required in order for the agent to control (hold,
transfer, conference, ) SIP Server calls.
Stream Manager is optional. Most of the Stream Manager capabilities
(music on hold, conference, and treatments) can be reproduced in Asterisk.
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Network
Inbound call
Asterisk
INVITE
NOTIFY
SIP Server
EventAgentNotReady
Genesys
Suite
EventAgentNotReady
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By staying in the signaling path, SIP Server is aware of any call status change and
can therefore produce call related events (EventRinging, EventEstabl i shed
,
EventReleased, ).
Any call control operation from the agent has to be performed using third party
call control (3pcc) procedure. In other words, agent desktop *must* be used for
any call control operation (beside the answer call operation). This includes but is
not limited to hold, transfer and conference requests.
The principle of contact center call is represented in the following picture.
Stream
Manager
Network
Inbound call
Asterisk
INVITE
INVITE
INVITE
SIP Server
EventEstablished
EventEstablished
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Genesys
Suite
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2 CALL FLOWS
2.1 SUBSCRIPTION
At startup, SIP Server sends subscription messages in order to be notified about
the endpoints status change.
Asterisk PBX provides NOTIFY messages to SIP Server according to the endpoints
status.
If the endpoints are not registered yet, Asterisk PBX reports their status as
closed.
Asterisk
SUBSCRIBE
NOTIFY
SIP Server
Asterisk
NOTIFY
SIP Server
REGISTER
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Network
Inbound call
Asterisk
INVITE
NOTIFY
SIP Server
EventAgentNotReady
As soon as the call is released from the endpoint, Asterisk notifies SIP Server
which then generates EventAgentReady so that the agent is now considered as
available for contact center calls.
See Annex B PRIVATE CALL.
Note: The exact same mechanism happens for private outbound calls. SIP Server
just sees NOTIFY message provided by Asterisk.
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Stream
Manager
Network
Inbound call
Asterisk
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INVITE
INVITE
SIP Server
EventRouteRequest
Genesys
Suite
Network
BYE
Asterisk
INVITE
INVITE
NOTIFY
SIP Server
RequestRouteCall
Genesys
Suite
EventAgentNotReady
EventRinging
EventEstablished
Then when the call is released, Asterisk notifies SIP Server with a NOTIFY message
just like for the case of private calls (EventAgentReady).
And because SIP Server is in the signaling path for that call, EventReleasedis also
generated.
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Network
Disconnect
Asterisk
BYE
BYE
BYE
NOTIFY
SIP Server
EventAgentReady
EventReleased
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Stream
Manager
INVITE
Asterisk
INVITE
NOTIFY
SIP Server
RequestMakeCall
INVITE
Then SIP Server contacts the requested destination number. For external
numbers, a rule shall be configured within SIP Server to dial out via Asterisk again
(see External access via Asterisk paragraph).
Once the destination answers the call, SIP Server disconnects ringback tone (BYE
to Stream Manager) and renegotiates with the agent endpoint (via Asterisk) so
that media stream is connected between the agent and the customer.
Stream
Manager
Network
Outbound call
Asterisk
BYE
INVITE
(RE)INVITE
SIP Server
EventEstablished
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Although disconnection would work if it were initiated directly from the agent
endpoint, it is good practice to always use desktop application in order to perform
any contact center call related action.
Therefore the disconnection is requested with RequestReleaseCal lto SIP Server.
SIP Server managing the 2 dialogs toward the agent and the customer is sending
BYE message to both of them and the call is eventually disconnected.
Network
Disconnect
Asterisk
BYE
BYE
BYE
NOTIFY
SIP Server
RequestReleaseCall
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3 CONFIGURATION
3.1 ENVIRONMENT
This chapter describes the following environment.
- Asterisk is connected to the network via a SIP gateway
- 2 SIP endpoints 2001 and 2002 are registered toward Asterisk
- Each endpoint is associated with a TLib desktop application
Gateway
SIP
Asterisk
SIP
SIP
SIP Server
SIP
TLIB
2002
TLIB
2001
3.2 GENESYS
3.2.1 SIP Server application
There are no particular configuration options related to Asterisk integration at SIP
Server application level.
3.2.2 Asterisk Trunk
The presence SUBSCRIBE/NOTIFY channel is configured by a DN of type Trunk.
The name choice for that DN is arbitrary.
Options needed for this Trunk DN are summarized in the following table.
Option (TServer section)
contact
Value
sip uri
subscribe-presence-domain
string
Version 0.2
Description
Indicates the host and SIP port where SIP Server shall
send SUBSCRIBE message. This is the Asterisk contact
in that case.
Domain name that is passed in SUBSCRIBE request
uri.
sip uri
subscribe-presence-expire
integer
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For example:
Value
sip uri
dual-dialog-enabled
false
make-call-rfc3725-flow
refer-enabled
1
false
reuse-sdp-on-reinvite
sip-hold-rfc3264
true
true
sip-initial-hold-rfc3264
true
subscribe-presence
string
For example:
Version 0.2
Description
Indicates the host and SIP port where SIP Server shall
send INVITE message to the endpoint. This is the
Asterisk contact in that case.
Consultation calls are handled using the same SIP
dialog toward Asterisk.
3pcc make call flow to be used according to RFC3725.
When using RFC3725 flow, REFER usage toward
Asterisk shall be disabled.
Never send a (RE)INVITE without SDP to Asterisk.
RTP stream hold is done using RFC3264 method
(sendonly).
RTP stream hold is done using RFC3264 method
(sendonly).
This is the name of the Trunk DN that is configured
for presence subscription messages toward Asterisk.
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Multiple rules can be defined. This part of the configuration is identical to the case
where SIP Server is deployed in standalone mode. Accesses to gateways are
replaced in this case by access to Asterisk.
3.3 ASTERISK
The following section describes configuration on the Asterisk side.
This is just an example of a possible Asterisk configuration and there may be
plenty of other ways to configure Asterisk.
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3.3.1 sip.conf
Two peers are configured describing both the gateway and SIP Server access.
For example:
[gwsim]
type=peer
host=10.0.0.1
port=5066
context=default
canreinvite=no
[gsip]
type=peer
username=gsip
host=10.0.0.1
context=default
canreinvite=no
Then each endpoint needs to be declared too. The user name of the endpoint shall
match the Extension DN configured on SIP Server side.
For example:
[2001]
type=friend
username=2001
host=dynamic
context=default
notifyringing=yes
canreinvite=no
[2002]
type=friend
username=2002
host=dynamic
context=default
notifyringing=yes
canreinvite=no
Note: SIP Server does not support receiving authentication challenges. For this
reason, Asterisk users must not be configured with secret option. If user
were configured with such option, Asterisk would challenge INVITE
messages issued by SIP Server on behalf the user and SIP Server would fail
responding to the challenge.
3.3.2 extensions.conf
On the dial plan side, each endpoint monitored by SIP Server shall contain a hint
entry. This is in order for Asterisk to properly accept presence subscription (from
SIP Server in that case) for those endpoints.
exten
exten
exten
exten
=>
=>
=>
=>
Version 0.2
2001,hint,SIP/2001
2001,1,Dial(SIP/2001,60)
2002,hint,SIP/2002
2002,1,Dial(SIP/2002,60)
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Version 0.2
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SIP Server
direct
NULL
TLib Clients
2400
2401
2001
Version 0.2
2002
Version 0.2
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<ep:activities><ep:away/></ep:activities>
</status></pp:person>
<note>Not online</note>
<tuple id="2001">
<contact priority="1">sip:2001@asterisk</contact>
<status><basic>closed</basic></status>
</tuple>
</presence>
[14] 200 OK ( To "Asterisk (1)" )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK293fe37a;rport;received=10.0.0.2
From: <sip:2001@asterisk>;tag=as2ead3322
To: <sip:gsip@phr:5060>;tag=EE9FF8B5-E97A-42BB-8B39-E3ABEADD569D-2
Call-ID: 1C4EAF69-820D-45B2-BA88-647EDFE47564-2@10.0.0.1
CSeq: 103 NOTIFY
Expires: 3600
Content-Length: 0
[18] RequestRegisterClient ( From "NULL" )
00:52:43.473 Trc 04541 RequestRegisterClient received from [1708]
message RequestRegisterClient
Att r i bu tePro toco lVers i on
' t se rver protoco l 4.2 '
AttributeApplicationName
'icom'
Att r i bu teSess ion ID
0
[19] EventLinkConnected ( To "NULL" )
@00:52:43.4730 [0] 7.5.000.11 send_to_client: message EventLinkConnected
AttributeApplicationName
'TServer-SIP-7.5'
AttributeSessionID
18284545
AttributeUserData
[2] 00 00..
AttributeRegistrationCode
0
AttributeEventSequenceNumber 0000000000000035
AttributeServerStartTime
45e542da00038658 (00:52:42.231000)
AttributeTimeinuSecs 473000
AttributeTimeinSecs
1172652763 (00:52:43)
[20] RequestQueryServer ( From "NULL" )
00:52:43.473 Trc 04541 RequestQueryServer received from [1708] (00020001 icom 10.0.0.1:3530)
message RequestQueryServer
AttributeReferenceID 18
AttributeExtensions
[2] 00 00..
[21] EventServerInfo ( To "NULL" )
@00:52:43.4730 [0] 7.5.000.11 send_to_client: message EventServerInfo
AttributeEventSequenceNumber 0000000000000036
AttributeTimeinuSecs 473000
AttributeTimeinSecs
1172652763 (00:52:43)
AttributeReferenceID 18
AttributeExtensions
[304] 00 02 00 00..
'T-Server'
'SIP Server, Version: 7.5.000.11 Compiled: Feb 19 2007 02:41:11
[22] RequestRegisterAddress (2400) ( From "2400" )
00:52:43.473 Trc 04541 RequestRegisterAddress received from [1708] (00020001 icom 10.0.0.1:3530)
message RequestRegisterAddress
AttributeReferenceID 19
AttributeExtensions
[2] 00 00..
AttributeAddressType 0 (Unknown)
AttributeControlMode 0
AttributeRegisterMode 0
AttributeThisDN '2400'
[23] EventRegistered (2400) ( To "2400" )
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Version 0.2
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Version 0.2
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SIP Server
TLib Clients
2001
Version 0.2
Version 0.2
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SIP Server
TLib Clients
2400
2001
Version 0.2
Version 0.2
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'4152540543'
Version 0.2
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Version 0.2
Version 0.2
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Version 0.2
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Version 0.2
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Version 0.2
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AttributeOtherDNRole 1
AttributeOtherDN
'4152540543'
[26] INVITE sip:4152540543@10.0.0.2 ( To "Asterisk (1)" )
INVITE sip:4152540543@10.0.0.2 SIP/2.0
From: <sip:2400@10.0.0.1>;tag=3DC47228-7872-44AF-A51C-116C4B171BDE-3
To: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
Call-ID: 6f5e2c9f289329eb41aecb10623f712e@10.0.0.2
CSeq: 1 INVITE
Content-Length: 206
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK982A37C6-43EA-4FF0-AB7D-1B86906D4ABB-8
Contact: <sip:10.0.0.1:5060>
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer
v=0
o=root 3025 3025 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 13512 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - [27] BYE sip:10.0.0.1:5080 ( To "Stream Manager (2)" )
BYE sip:10.0.0.1:5080 SIP/2.0
From: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
To: <sip:annc@phr:5080;play=mymusic/nightvision>;tag=A1BEC915-4E42-416B-A628-1D0E60F95AC2-4
Call-ID: 04C499D8-6173-48C2-830E-E89C59B7B074-3@10.0.0.1
CSeq: 3 BYE
Content-Length: 0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK982A37C6-43EA-4FF0-AB7D-1B86906D4ABB-9
[28] 200 OK ( From "Stream Manager (2)" )
SIP/2.0 200 OK
From: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
To: <sip:annc@phr:5080;play=mymusic/nightvision>;tag=A1BEC915-4E42-416B-A628-1D0E60F95AC2-4
Call-ID: 04C499D8-6173-48C2-830E-E89C59B7B074-3@10.0.0.1
CSeq: 3 BYE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK982A37C6-43EA-4FF0-AB7D-1B86906D4ABB9;received=10.0.0.1
Contact: <sip:10.0.0.1:5080>
Content-Length: 0
[29] 200 OK ( From "Asterisk (1)" )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK982A37C6-43EA-4FF0-AB7D-1B86906D4ABB8;received=10.0.0.1
From: <sip:2400@10.0.0.1>;tag=3DC47228-7872-44AF-A51C-116C4B171BDE-3
To: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
Call-ID: 6f5e2c9f289329eb41aecb10623f712e@10.0.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4152540543@10.0.0.2>
Content-Type: application/sdp
Content-Length: 271
v=0
o=root 3025 3027 IN IP4 192.168.1.201
Version 0.2
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s=session
c=IN IP4 192.168.1.201
t=0 0
m=audio 34970 RTP/AVP 0 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - [30] ACK sip:4152540543@10.0.0.2 ( To "Asterisk (1)" )
ACK sip:4152540543@10.0.0.2 SIP/2.0
From: <sip:2400@10.0.0.1>;tag=3DC47228-7872-44AF-A51C-116C4B171BDE-3
To: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
Call-ID: 6f5e2c9f289329eb41aecb10623f712e@10.0.0.2
CSeq: 1 ACK
Content-Length: 0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK982A37C6-43EA-4FF0-AB7D-1B86906D4ABB-8
Call-Info: <http://genesyslab.com>; 04C499D8-6173-48C2-830E-E89C59B7B074-4%4010.0.0.1;genrt=as21836576;gen-lt=as370214c9
[31] ACK sip:2001@10.0.0.2 ( To "Asterisk (1)" )
ACK sip:2001@10.0.0.2 SIP/2.0
From: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
To: sip:2400@10.0.0.1;tag=as21836576
Call-ID: 04C499D8-6173-48C2-830E-E89C59B7B074-4@10.0.0.1
CSeq: 1 ACK
Content-Length: 271
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK982A37C6-43EA-4FF0-AB7D-1B86906D4ABB-7
Call-Info: <http://genesyslab.com>; 6f5e2c9f289329eb41aecb10623f712e%4010.0.0.2;genrt=as370214c9;gen-lt=3DC47228-7872-44AF-A51C-116C4B171BDE-3
v=0
o=root 3025 3027 IN IP4 192.168.1.201
s=session
c=IN IP4 192.168.1.201
t=0 0
m=audio 34970 RTP/AVP 0 110 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - [32] BYE sip:10.0.0.1:5060 ( From "Asterisk (1)" )
BYE sip:10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=z9hG4bK09fcb3cc;rport
From: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
To: <sip:2400@10.0.0.1>;tag=3DC47228-7872-44AF-A51C-116C4B171BDE-3
Contact: <sip:4152540543@10.0.0.2>
Call-ID: 6f5e2c9f289329eb41aecb10623f712e@10.0.0.2
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[33] BYE sip:2001@10.0.0.2 ( To "Asterisk (1)" )
BYE sip:2001@10.0.0.2 SIP/2.0
From: "4152540543" <sip:4152540543@10.0.0.2>;tag=as370214c9
To: sip:2400@10.0.0.1;tag=as21836576
Call-ID: 04C499D8-6173-48C2-830E-E89C59B7B074-4@10.0.0.1
CSeq: 3 BYE
Version 0.2
Version 0.2
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Version 0.2
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SIP Server
TLib Clients
2001
SIP-7.5::
Stream Manager
Version 0.2
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Version 0.2
Version 0.2
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SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5191;received=10.0.0.1
From: <sip:04152540543@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-29
To: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2001@10.0.0.2>
Content-Type: application/sdp
Content-Length: 150
v=0
o=root 2983 2983 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 15480 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - [10] INVITE sip:04152540543@colinux:5060 ( To "Asterisk" )
INVITE sip:04152540543@colinux:5060 SIP/2.0
From: <sip:2001@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-30
To: <sip:04152540543@10.0.0.1:5060>
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-42@10.0.0.1
CSeq: 1 INVITE
Content-Length: 150
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-192
Contact: <sip:10.0.0.1:5060>
Call-Info: <http://genesyslab.com>; F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41%4010.0.0.1;genrt=as16f1101c;gen-lt=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-29
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: 100rel,timer
v=0
o=root 2983 2983 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 15480 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - [11] 100 Trying ( From "Asterisk" )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5192;received=10.0.0.1
From: <sip:2001@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-30
To: <sip:04152540543@10.0.0.1:5060>
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-42@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:04152540543@10.0.0.2>
Content-Length: 0
[12] 180 Ringing ( From "Asterisk" )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5192;received=10.0.0.1
Version 0.2
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From: <sip:2001@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-30
To: <sip:04152540543@10.0.0.1:5060>;tag=as5b53fb1a
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-42@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:04152540543@10.0.0.2>
Content-Length: 0
[13] EventNetworkReached (2001,04152540543) ( To "2001" )
@13:08:02.4030 [0] 7.5.000.14 distribute_event: message EventNetworkReached
AttributeEventSequenceNumber 00000000000000d8
AttributeTimeinuSecs 403000
AttributeTimeinSecs
1173989282 (13:08:02)
AttributeExtensions
[23] 00 01 01 00..
'BusinessCall' 0
AttributeANI '2001'
AttributeDNIS '04152540543'
AttributeUserData
[39] 00 03 00 00..
'KEY1' 'ABCD'
'KEY2' 1234
'KEY3' bin: 01 02
AttributeCallUUID
'G4Q0S4JBQ13ML76VQK9CCPSQ48000018'
AttributeConnID 0117016f33fb000e
AttributeCallID 14
AttributeCallType
3
AttributeCallState
0
AttributeThisDNRole
1
AttributeAgentID
'6001'
AttributeThisDN '2001'
AttributeOtherDNRole 2
AttributeOtherDN
'04152540543'
[14] INVITE sip:annc@phr:5080;play=music/ring_back ( To "Stream Manager" )
INVITE sip:annc@phr:5080;play=music/ring_back SIP/2.0
From: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
To: <sip:annc@phr:5080;play=music/ring_back>
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-43@10.0.0.1
CSeq: 1 INVITE
Content-Length: 150
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-193
Contact: <sip:10.0.0.1:5060>
Max-Forwards: 70
v=0
o=root 2983 2983 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 15480 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - [15] 200 OK ( From "Stream Manager" )
SIP/2.0 200 OK
From: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
To: <sip:annc@phr:5080;play=music/ring_back>;tag=6ACCDD30-E25F-4087-8188-B2699B40D4C0-13
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-43@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5193;received=10.0.0.1
Contact: <sip:172.21.9.220:5080>
Content-Type: application/sdp
Content-Length: 152
Version 0.2
46/50
v=0
o=Genesys 13 13 IN IP4 172.21.9.220
s=StreamManager 7.5.004.02 play
c=IN IP4 172.21.9.220
t=0 0
m=audio 20026 RTP/AVP 0
a=rtpmap:0 pcmu/8000
[16] ACK sip:2001@10.0.0.2 ( To "Asterisk" )
ACK sip:2001@10.0.0.2 SIP/2.0
From: <sip:04152540543@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-29
To: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41@10.0.0.1
CSeq: 1 ACK
Content-Length: 152
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-191
v=0
o=Genesys 13 13 IN IP4 172.21.9.220
s=StreamManager 7.5.004.02 play
c=IN IP4 172.21.9.220
t=0 0
m=audio 20026 RTP/AVP 0
a=rtpmap:0 pcmu/8000
[17] ACK sip:172.21.9.220:5080 ( To "Stream Manager" )
ACK sip:172.21.9.220:5080 SIP/2.0
From: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
To: <sip:annc@phr:5080;play=music/ring_back>;tag=6ACCDD30-E25F-4087-8188-B2699B40D4C0-13
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-43@10.0.0.1
CSeq: 1 ACK
Content-Length: 0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-193
[18] 200 OK ( From "Asterisk" )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5192;received=10.0.0.1
From: <sip:2001@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-30
To: <sip:04152540543@10.0.0.1:5060>;tag=as5b53fb1a
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-42@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:04152540543@10.0.0.2>
Content-Type: application/sdp
Content-Length: 150
v=0
o=root 2983 2983 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 15580 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - [19] INVITE sip:2001@10.0.0.2 ( To "Asterisk" )
INVITE sip:2001@10.0.0.2 SIP/2.0
From: <sip:04152540543@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-29
To: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41@10.0.0.1
CSeq: 2 INVITE
Content-Length: 150
Version 0.2
47/50
Content-Type: application/sdp
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-194
Contact: <sip:10.0.0.1:5060>
Call-Info: <http://genesyslab.com>; F20C79BA-EF95-42E0-97FC-E3ED5AD64404-42%4010.0.0.1;genrt=as5b53fb1a;gen-lt=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-30
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: timer
v=0
o=root 2983 2983 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 15580 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - [20] BYE sip:172.21.9.220:5080 ( To "Stream Manager" )
BYE sip:172.21.9.220:5080 SIP/2.0
From: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
To: <sip:annc@phr:5080;play=music/ring_back>;tag=6ACCDD30-E25F-4087-8188-B2699B40D4C0-13
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-43@10.0.0.1
CSeq: 2 BYE
Content-Length: 0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-195
[21] 200 OK ( From "Stream Manager" )
SIP/2.0 200 OK
From: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
To: <sip:annc@phr:5080;play=music/ring_back>;tag=6ACCDD30-E25F-4087-8188-B2699B40D4C0-13
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-43@10.0.0.1
CSeq: 2 BYE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5195;received=172.21.9.220
Contact: <sip:172.21.9.220:5080>
Content-Length: 0
[22] 200 OK ( From "Asterisk" )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5194;received=10.0.0.1
From: <sip:04152540543@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-29
To: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41@10.0.0.1
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2001@10.0.0.2>
Content-Type: application/sdp
Content-Length: 150
v=0
o=root 2983 2984 IN IP4 10.0.0.2
s=session
c=IN IP4 10.0.0.2
t=0 0
m=audio 15480 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - [23] ACK sip:2001@10.0.0.2 ( To "Asterisk" )
ACK sip:2001@10.0.0.2 SIP/2.0
From: <sip:04152540543@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-29
Version 0.2
48/50
To: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41@10.0.0.1
CSeq: 2 ACK
Content-Length: 0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-194
[24] ACK sip:04152540543@10.0.0.2 ( To "Asterisk" )
ACK sip:04152540543@10.0.0.2 SIP/2.0
From: <sip:2001@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-30
To: <sip:04152540543@10.0.0.1:5060>;tag=as5b53fb1a
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-42@10.0.0.1
CSeq: 1 ACK
Content-Length: 0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-192
[25] EventEstablished (2001,04152540543) ( To "2001" )
@13:08:04.4560 [0] 7.5.000.14 distribute_event: message EventEstablished
AttributeEventSequenceNumber 00000000000000d9
AttributeTimeinuSecs 456000
AttributeTimeinSecs
1173989284 (13:08:04)
AttributeExtensions
[106] 00 03 01 00..
'WrapUpTime' 0
'SIP-Call-ID' 'F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41@10.0.0.1'
'BusinessCall' 0
AttributeANI '2001'
AttributeDNIS '04152540543'
AttributeUserData
[39] 00 03 00 00..
'KEY1' 'ABCD'
'KEY2' 1234
'KEY3' bin: 01 02
AttributeCallUUID
'G4Q0S4JBQ13ML76VQK9CCPSQ48000018'
AttributeConnID 0117016f33fb000e
AttributeCallID 14
AttributeCallType
3
AttributeCallState
0
AttributeThisDNRole
1
AttributeAgentID
'6001'
AttributeThisDN '2001'
AttributeOtherDNRole 2
AttributeOtherDN
'04152540543'
[26] RequestReleaseCall (2001) ( From "2001" )
13:08:08.302 Trc 04541 RequestReleaseCall received from [1640] (00040003 DesktopApp 10.0.0.1:1729)
message RequestReleaseCall
AttributeReferenceID 38
AttributeConnID 0117016f33fb000e
AttributeThisDN '2001'
[27] BYE sip:2001@10.0.0.2 ( To "Asterisk" )
BYE sip:2001@10.0.0.2 SIP/2.0
From: <sip:04152540543@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-29
To: <sip:2001@10.0.0.1:5060>;tag=as16f1101c
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-41@10.0.0.1
CSeq: 3 BYE
Content-Length: 0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK9C92C5D9-4B03-45E4-86EC-AB2BEA98FAE5-196
[28] BYE sip:04152540543@10.0.0.2 ( To "Asterisk" )
BYE sip:04152540543@10.0.0.2 SIP/2.0
From: <sip:2001@10.0.0.1:5060>;tag=506F7777-63E1-4BC9-858E-AFDE67B2AFCB-30
To: <sip:04152540543@10.0.0.1:5060>;tag=as5b53fb1a
Call-ID: F20C79BA-EF95-42E0-97FC-E3ED5AD64404-42@10.0.0.1
CSeq: 2 BYE
Content-Length: 0
Version 0.2
Version 0.2
49/50
Version 0.2
50/50