SIP Trunk 2 User Manual

SIP Trunk 2 IP-PBX User Guide
（Asterisk）

Ver1.1.1 2017/11/21

Index

1.

SIP Trunk 2 Overview ………………………………………………………

3

2. Purchase/Settings in Web Portal ……………………………… 5

3. Configuration Example of your IP-PBX ……………………………… 12

4. Technical Data ……………………………… 24

2

SIP Trunk 2 is a next generation IP phone service that connects to PBX making an external line call
which is compatible to Asterisk, Aspire X IP-PBX.

１．SIP Trunk 2 Overview

＜SIP Trunk 2 FEATURE HIGHLIGHTS＞

■ Compatible to Asterisk, Aspire X PBX.

■ Options for “ Authentication Method” are:

• Password Authentication

•

Authentication with IP Address

• Authentication using both IP Address and Password.

■ CPS （Call Per Second) has been significantly improved from normal SIP trunk.
*Our Cloud PBX Recording Option is currently not supported by SIP trunk 2
(If you need the recording option, please Contact us)

===== Verified IP-PBX =====
・Asterisk

Asterisk PBX/1.4.x
Asterisk PBX 1.6.x
Asterisk PBX 1.8.x

Asterisk PBX 11

Asterisk PBX 12

・Aspire X

IP3WW-32VOIPDB-A1
version: 05.01

*IP-PBX versions not listed above are not fully supported by SIP trunk 2.
========================

※Please permit on your firewall incoming network traffic from our VoIP server IP addresses with
5060, 10000~20000 UDP ports.

3

１．SIP Trunk 2 Overview

[Sample Configuration SIP trunk2 with IP-PBX]

*If SIP trunk2 Unique was purchased BEFORE March 9, 2017,

Unique: 0000123456

SIP Trunk2 Server: xxx.xxx.xxx.xxx
Your IP-PBX : IP Address 000.000.000.000
DIDs: 0312345678 , 0312123434

*If SIP trunk2 Unique was purchased BEFORE March 9, 2017,

In case of Japanese toll free numbers such as prefix 0120,
0800 and 0570, you should set its background number
showing in Phone Number List of the web portal.

ex.) A number enclosed in parentheses is its background

number.

0120****** [03******]
*Customers who use multiple SIP trunk 2 Unique together

( the one purchased before Mar, 9 2017 and the one
purchased Mar, 10 2017 afterwards) can check each SIP

trunk2 unique setting on the SIP trunk2 Unique Details page

as below

Checked : Back ground number
Unchecked : Free Call or Navi-Dial number

4

１．SIP Trunk 2 Overview

xxx.xxx.xxx.xxx

SIP Trunk 2

To:<

sip:

0312345678

@

000.000.000.000

>

Recipient number is set “To header” and “Alert­Into” in SIP messages for Incoming call.
See section 4 ”Technical Data" for more details.

From: <

sip:

0312123434

@xxx.xxx.xxx.xxx

>

Caller ID must be set “From header” for outgoing call.
See section 4 ”Technical Data" for more details.

Your IP-PBX

0000.0000.0000.0000

Ext. 200

Ext. 201

DID: 0312123434

DID: 0312345678

Ext. 200

Ext. 201

Image 1. Configuration Diagram of Incoming/Outgoing Calls

Call TO in case DID : “0312345678”, ext. 200 will ring.

5

Call TO in case DID : “

0312123434

”,

ext

201 will ring.

Outgoing call from ext. 200, “0312345678” is set as caller ID
Outgoing call from ext. 201, “

0312123434” is set as caller ID.

２．Purchase/Settings in Web Portal

For purchasing SIP Trunk 2, access the UI of our IP-PBX.

Buy additional SIP trunk channel for 2 or more simultaneous external calls.

＜SIP Trunk 2 Purchase Screen>

①

②

① Select “Purchase” at the top menu and choose ”Purchase Unique”

③

in

Circle Management Page

② Select quantity of SIP trunk 2
③ Click “Add to Cart” to proceed for your purchase

6

２．Purchase/Settings in Web Portal

Purchase phone number here
*At least one phone number will be needed for external phone calls through SIP Trunk

＜

Phone Number Purchase Screen>

②

①

① Select “Purchase” at the top menu and choose

”Purchase Phone Number”

in Circle Management Page

② On the Purchase Phone Number page, find your desired phone number by
clicking “Search” button. Add to cart and select “Your Cart” to proceed.

7

２．Purchase/Settings in Web Portal

＜SIP Trunk 2 List＞

①

②

③

① Select “SIP Trunk List” to open all your SIP trunk account
② Select the icon under “Detail” for detailed settings of SIP Trunk

(See next page)

③

Your unique is used as client user ID of your user PBX end

8

２．Purchase/Settings in Web Portal

＜SIP Trunk 2 Detailed Settings ・ Password Authentication＞

①
②

③
④

⑤

⑥
⑦
⑧
⑨

① Login server name of SIP Trunk 2

⑩

②

SIP Server IP Address

Please configure it as [peer] in sip.conf on your Asterisk.

*Please refer p.14 for details.
③ Unique is used as client user ID of your user PBX end.
④ Item “Name” is where you can name/rename your SIP Trunk account.

⑤

Select authentication

method

as “Password Authentication”

⑤

Select authentication

method

as “Password Authentication”

⑥ Enter your terminal password is used as client user password of your PBX end.
⑦ In case you use Free Call or Navi-Dial number, please designate format of

incoming number configured on your SIP server.
Checked : Please configure 0ABJ number
Unchecked : Please configure Free Call or Navi-Dial number.

(* Changing check box status is not allowed.)

⑧ Set multiple call count. It’s 1 by default. Purchase “Additional 1 channel for

SIP Trunk 2” if you need more than 2 concurrent calls.

⑨ If this option is checked, you can specify channel reservation for multiple

outbound calls.

⑩ If this option is checked, you can specify channel reservation for multiple

9

inbound

calls.

*Note ⑨ and ⑩

The total channel reservation (outbound + inbound) must not exceed the multiple call count.
Please refer p.12 for further details.

２．Purchase/Settings in Web Portal

＜SIP Trunk 2 Detailed Settings ・ Authentication with IP Address＞

①

②
③

④
⑤

⑥

⑦

⑧
⑨

⑩
⑪

① Login server name of SIP Trunk 2
② Our SIP Server IP Address

Please configure it as [peer] in sip.conf on your Asterisk.

*Please refer

p.14

for details

③ Unique is used as client user ID of your user PBX end.
④ Item “Name” is where you can name/rename your SIP Trunk account.
⑤ Select authentication method as “Authentication with IP Address”
⑥ Enter a public IP address / a port number of your IP-PBX *You can add multiple IP

addresses/ports from “+Insert” button.

⑦

Your IP

-

PBX will receive incoming call if ticked. *If

unticked

it will work only for

outgoing calls.

⑧ In case you use Free Call or Navi-Dial number, please designate format of

incoming number configured on your SIP server.
Checked : Please configure 0ABJ number
Unchecked : Please configure Free Call or Navi-Dial number.

(

* Changing check box status is not allowed.

)

⑨ Set multiple call count. It’s 1 by default. Purchase “Additional 1 channel for

SIP Trunk 2” if you need more than 2 concurrent calls.

⑩ If this option is checked, you can specify channel reservation for multiple

outbound calls.

⑪

If

this option is checked, you can specify channel reservation for multiple inbound

10

⑪

If

this option is checked, you can specify channel reservation for multiple inbound

calls.

*Note ⑩ and ⑪

The total channel reservation (outbound + inbound) must not exceed the multiple call count.

Please refer p.12 for further details.

２．Purchase/Settings in Web Portal

＜SIP Trunk 2 Detailed Settings ・

Authentication using both IP Address and Password>

①
②

④

③

⑤

⑥
⑦

⑧
⑨
⑩

⑪

①

Login server name of SIP Trunk 2

② Our SIP Server IP Address

Please configure it as [peer] in sip.conf on your Asterisk.

*Please refer p.14 for details

③ Unique is used as client user ID of your user PBX end.
④ Item “Name” is where you can name/rename your SIP Trunk account.
⑤

Select authentication method as “Authentication

using Both IP Address and Password”

Select authentication method as “Authentication

using Both IP Address and Password”

⑥ Enter your terminal password is used as client user password of your PBX end.
⑦ Enter a public IP address / a port number of your IP-PBX *You can add multiple IP addresses/ports

from “+Insert” button.

⑧ In case you use Free Call or Navi-Dial number, please designate format of incoming number

configured on your SIP server.

Checked : Please configure 0ABJ number

Unchecked

: Please configure Free Call or

Navi

-

Dial number.

(* Changing check box status is not allowed.)

⑨ Set multiple call count. It’s 1 by default. Purchase “Additional 1 channel for

SIP Trunk 2” if you need more than 2 concurrent calls.
⑩ If this option is checked, you can specify channel reservation for multiple outbound calls.
⑪ If this option is checked, you can specify channel reservation for multiple inbound calls.

11

*

Note

⑩

and

⑪

The total channel reservation (outbound + inbound) must not exceed the multiple call count.
Please refer p.12 for further details.

２．Purchase/Settings in Web Portal

＜SIP Trunk 2 Detailed Settings ・

Channel Reservation Settings>

This section briefly discusses sample settings for SIP trunk2 channel reservations:

①

Not using channel reservation (default)

If Multiple call count = 4, no reservations.

This means outbound and inbound channels maximizes all channel limit.
There is a possibility of all channels will be used for outbound calls only or inbound calls only.

②

Using channel reservation for outbound call

If

Multiple call count = 4

,

If Channel reservation for outbound call = 1

This would mean that the selected unique will have:

Allowed Outbound Calls = 4
Allowed Inbound Calls = 3

If Multiple call count = 4,
If Channel reservation for outbound call = 4

This would mean that the selected unique will have:

Allowed Outbound Calls = 4

Allowed

Inbound Calls =

0

Allowed

Inbound Calls =

0

③

Using channel reservation for inbound call

If Multiple call count = 4,
If Channel reservation for inbound call = 3

This would mean that the selected unique will have:

Allowed Outbound Calls = 1
Allowed Inbound Calls = 4

If Multiple call count = 4,
If Channel reservation for inbound call = 4

12

This would mean that the selected unique will have:

Allowed Outbound Calls = 0
Allowed Inbound Calls = 4

２．Purchase/Settings in Web Portal

＜SIP Trunk 2 Detailed Settings ・

Channel Reservation Settings>

④

U

sing both

outbound

and inbound

channel reservation

④

U

sing both

outbound

and inbound

channel reservation

If Multiple call count = 4,
If Channel reservation for outbound call = 3
If Channel reservation for inbound call = 1

This would mean that the selected unique will have:

Allowed

Outbound Calls =

3

Allowed Inbound Calls = 1

13

２．Purchase/Settings in Web Portal

Select phone number(s) you desire to assign to SIP Trunk 2

＜

Phone Number List>

②

①

① Click “Phone Number List” to open your Phone Number List.
② Select SIP Trunk 2 unique for phone number(s) you desire to assign for it

14

３．Configuration Example of your IP-PBX

3.1. Configuration Example in Asterisk

[Account Example]

Unique: 0000123456

Password: password

DIDs: 0312345678 , 0312123434
Extensions: 200, 201
Login Server: xxx.xxx.xxx.xxx
※login the web portal to confirm your login server.

[Settings Example]

Incoming call for 0312345678 is to be arrived at Ext. 200.
Incoming call for 0312123434 is to be arrived at Ext. 201.

Outgoing call from a phone with Ext. 200 is to be called with CallerID: 0312345678
Outgoing call from a phone with Ext. 201 is to be called with CallerID: 0312123434

; ------------------

;

sip.conf

(for either password or IP address with password authentication)

; ------------------

[general]
allowguest=no
maxexpirey=3600
defaultexpirey=3600

port=5060

bindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
language=jp

register =>

0000123456

:password@siptr

register =>

0000123456

:password@siptr

[siptr]
type=friend
username=0000123456
secret=password
context=inbound
canreinvite=no

host=

xxx.xxx.xxx.xxx

insecure=port,invite
disallow=all
allow=ulaw
qualify=yes
nat=yes
;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11

15

;<see also next page for the rest settings of sip.conf>

３．Configuration Example of your IP-PBX

; -----------------­; sip.conf (for either password or IP address with password authentication)
; ------------------

[

200

]

type=friend
username=200

secret=200pass
host=dynamic
context=outbound-1

[

201

]

type=friend
username=201
secret=201pass
host=dynamic
context=outbound-2

;<see also next page for sip.conf for IP address authentication>

16

３．Configuration Example of your IP-PBX

; -----------------­; sip.conf (for IP address authentication)
; ------------------

[general]

allowguest=no
maxexpirey=3600
defaultexpirey=3600
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all

allow=

ulaw

allow=

ulaw

language=jp
[siptr]

type=friend
context=inbound
canreinvite=no

host=xxx.xxx.xxx.xxx

insecure

=port,invite

insecure

=port,invite

disallow=all
allow=ulaw
qualify=yes

nat=yes

;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11

[peer1]

type=friend

context=inbound
Host=xxx.xxx.xxx.xxx

nat=yes

[peer2]
type=friend
context=inbound

Host=

xxx.xxx.xxx.xxx

nat=yes

;please refer to P.10 ② for checking host IP address to be configured as peer.

[200]
type=friend
username=200

secret=

200

pass

host=dynamic
context=outbound-1

[201]
type=friend
username=201
secret=201pass

17

host=dynamic

context=outbound-2

３．Configuration Example of your IP-PBX

; -----------------­; extensions.conf
; ------------------

[general]

writeprotect

=no

priorityjumping=yes

[inbound]
exten => 0312345678,1, Dial(SIP/200,120,t)
exten => 0312345678,2,Congestion
exten => 0312345678,102,Busy

exten => 0312123434,1, Dial(SIP/201,120,t)
exten => 0312123434,2,Congestion
exten => 0312123434,102,Busy

[outbound-1]
exten => _0., 1,Set(CALLERID(num)= 0312345678
exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T)

exten

=> _0., 3,Congestion

exten => _0.,104,Busy

exten => _1., 1,Set(CALLERID(num)= 0312345678
exten => _1., 2,Dial(SIP/${EXTEN}@siptr,120,T)
exten => _1., 3,Congestion
exten => _1.,104,Busy

;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on.

exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T)
exten => _ XXX, 2,Congestion
exten => _ XXX, 102,Busy
; XXX represents 3 digit-extensions. Please adjust digit number as yours.

;<see also next page for the rest settings of extensions.conf>

18

３．Configuration Example of your IP-PBX

[outbound-2]
exten => _0., 1,Set(CALLERID(num)= 0312123434)
exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T)

exten

=> _0., 3,Congestion

exten => _0.,104,Busy

exten => _1., 1,Set(CALLERID(num)= 0312123434)
exten => _1., 2,Dial(SIP/${EXTEN}@siptr,120,T)
exten => _1., 3,Congestion
exten => _1.,104,Busy

;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on.

exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T)
exten => _ XXX, 2,Congestion
exten => _ XXX, 102,Busy
; XXX represents 3 digit-extensions. Please adjust digit number as yours.

19

３．Configuration Example of your IP-PBX

3.2. Configuration Example to limit multiple call count for each extension group in Asterisk.

[Settings Example]

Group １:

Max multiple count

2

Extensions

201 ～202

Phone Numbers

03-1234-5678

Group

２

:

Max multiple count

3

Extensions

Set max multiple call count (for external calls) as 2 for Group 1

Set max multiple call count (for external calls) as 3 for Group 2

Group

２

:

Extensions

301

～

302

Phone Numbers

03-1212-3434

; -----------------­; sip.conf (for either password or IP address with password authentication)
; ------------------

[general]
allowguest=no
maxexpirey=3600
defaultexpirey=3600
context=extd
port=5060

bindaddr

=0.0.0.0

srvlookup=yes
disallow=all
allow=ulaw
language=jp

register=>0000123456:[e-mail]/0000123456

[

0000123456

]

type=friend
username=0000123456
secret=password
host=xxx.xxx.xxx.xxx
insecure=port,invite
context=inbound

qualify=

yes

nat=yes
;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11

;<see also next page for the rest settings of sip.conf>

20

３．Configuration Example of your IP-PBX

; -----------------­; sip.conf (for either password or IP address with password authentication)
; ------------------

; Group 1

[201]
type=friend
context=group1_outbound
username=201
secret=password
host=dynamic

[202]
type=friend
context=group1_outbound
username=202
secret=password
host=dynamic

; Group 2
[301]
type=friend
context=group2_outbound
username=301

secret=password

host=dynamic

[302]
type=friend
context=group2_outbound
username=302

secret=password

secret=password

host=dynamic

;<see also next page for sip.conf for IP address authentication>

21

３．Configuration Example of your IP-PBX

;-------------­;sip.conf (IP address authentication)
;--------------

[general]

allowguest

=no

maxexpirey=3600
defaultexpirey=3600
context=extd
port=5060
bindaddr=0.0.0.0
srvlookup=yes
disallow=all

allow=

ulaw

language=jp

[siptr]
type=friend
context=inbound
canreinvite=no

host= xxx.xxx.xxx.xxx

insecure=

port,invite

insecure=

port,invite

disallow=all
allow=ulaw
qualify=yes

nat=yes

;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11

[peer1]

[peer1]

type=friend
context=inbound
Host=xxx.xxx.xxx.xxx

nat=yes

[peer2]
type=friend
context=inbound

Host=

xxx.xxx.xxx.xxx

nat=yes

;please refer to P.10 ② for checking host IP address to be configured as peer.

;<see also next page for the rest settings of sip.conf>

22

３．Configuration Example of your IP-PBX

;-------------­;sip.conf (IP address authentication)
;--------------

; Group 1

[201]
type=friend
context=group1_outbound
username=201
secret=password
host=dynamic

[

202

]

[

202

]

type=friend
context=group1_outbound
username=202
secret=password
host=dynamic

; Group 2
[301]
type=friend
context=group2_outbound
username=301
secret=password
host=dynamic

[

302

]

type=friend
context=group2_outbound
username=302
secret=password
host=dynamic

23

３．Configuration Example of your IP-PBX

<extensions.conf Example in your Asterisk>

; ------------------

;

extensions.conf

;

extensions.conf

;

------------------

[general]
writeprotect=no
priorityjumping=yes

; Group 1

[inbound]
exten => 0312345678,1,NoOp(EXTEN: ${EXTEN})
exten => 0312345678,2,Set(GROUP(CALLS)=GROUP1)
exten => 0312345678,3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)})
exten => 0312345678,4,Set(MAXCALLS=2)
exten => 0312345678,5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup)

exten

=>

0312345678

,6,Dial(SIP/

201

&SIP/

202

,120

)

exten => 0312345678,7,Congestion
exten => 0312345678,106,Busy

; Group 2
exten => 0312123434,1,NoOp(EXTEN: ${EXTEN})
exten => 0312123434,2,Set(GROUP(CALLS)=GROUP2)
exten => 0312123434,3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)})

exten

=>

0312123434

,4,Set(MAXCALLS=

3

)

exten => 0312123434,5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup)
exten => 0312123434,6,Dial(SIP/301&SIP/302,120)
exten => 0312123434,7,Congestion
exten => 0312123434,106,Busy

24

;<see also next page for the rest settings of

extensions.conf

>

３．Configuration Example of your IP-PBX

<extensions.conf Example in your Asterisk>

; Group 1
[group1_outbound]

exten

=> _0., 1,Set(CALLERID(num)=

0312345678

)

exten => _0., 2,Set(CALLERID(name)=GROUP1)
exten => _0., 3,Set(GROUP(CALLS)=GROUP1)
exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)})
exten => _0., 5,Set(MAXCALLS=2)
exten => _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup)
exten => _0., 7,Dial(SIP/${EXTEN}@0000123456,120)

exten

=> _0., 8,Congestion

exten

=> _0.,106,Busy

exten => _1., 1,Set(CALLERID(num)=0312345678)
exten => _1., 2,Set(CALLERID(name)=GROUP1)
exten => _1., 3,Set(GROUP(CALLS)=GROUP1)
exten => _1., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)})
exten => _1., 5,Set(MAXCALLS=2)

exten

=> _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?

Hangup

)

exten

=> _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?

Hangup

)

exten => _1., 7,Dial(SIP/${EXTEN}@0000123456,120)
exten => _1., 8,Congestion exten => _0.,106,Busy

exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T)
exten => _ XXX, 2,Congestion
exten => _ XXX, 102,Busy

; Group 2

[group2_outbound]
exten => _0., 1,Set(CALLERID(num)= 0312123434)
exten => _0., 2,Set(CALLERID(name)=GROUP2)
exten => _0., 3,Set(GROUP(CALLS)=GROUP2)
exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)})
exten => _0., 5,Set(MAXCALLS=3)

exten

=> _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?

Hangup

)

exten => _0., 7,Dial(SIP/${EXTEN}@0000123456,120)
exten => _0., 8,Congestion
exten => _0.,106,Busy

exten => _1., 1,Set(CALLERID(num)= 0312123434)
exten => _1., 2,Set(CALLERID(name)=GROUP2)
exten => _1., 3,Set(GROUP(CALLS)=GROUP2)

exten

=> _1., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)})

exten => _1., 5,Set(MAXCALLS=3)
exten => _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup)
exten => _1., 7,Dial(SIP/${EXTEN}@0000123456,120)
exten => _1., 8,Congestion
exten => _1.,106,Busy

25

exten

=> _ XXX, 1,Dial(SIP/${EXTEN},120,T)

exten => _ XXX, 2,Congestion
exten => _ XXX, 102,Busy

４．Technical Data

4.1. SIP REGISTER message

：

■ Sending REGISTER message Is required to register your ID, IP address and port number for authentication.

REGISTER

your IP-PBX

000.000.000.000

SIP Trunk 2

xxx.xxx.xxx.xxx

Your ID (SIP Trunk 2 unique
number

IP address of
SIP Trunk 2

From:<sip

:

0000123456

@

xxx.xxx.xxx.xxx>;

tag=as

04bc6a95

To: <sip: [e-mail]>
Call-ID: [email protected]

1

100 Trying
From: <sip: [e-mail]>;tag=as04bc6a95
To: <sip: [e-mail]>
Call-ID: [email protected]

2

401 Unauthorized
From: <sip: [e-mail]>;tag=as04bc6a95
To: <sip: [e-mail]>;tag=as245298a3
Call-ID: [email protected]

3

REGISTER(with credential information)
From: <sip: [e-mail]>;tag=as2031f6e2
To: <sip: [e-mail]>

Call-ID:34d61b985ef9d9c12d819a9c5549471

f@

127.0.0.1

4

SIP/2.0 100 Trying
From: <sip: [e-mail]>;tag=as2031f6e2
To: <sip: [e-mail]>
Call-ID: [email protected]

5

200 OK

From:<sip:0000123456

@

xxx.xxx.xxx.xxx>;

tag=as

2031f6e2

figure 4.1 SIP flow for REGISTER

From:<sip

:

0000123456

@

xxx.xxx.xxx.xxx>;

tag=as

2031f6e2

To: <sip: [e-mail]>;tag=as245298a3
Call-ID: [email protected]

6

※

Sending REGISTER message is NOT required in case your authentication method is “Authentication

with IP Address”

26

４．Technical Data

4.1.1 PBX → GUEST

REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;rport

From: <sip:

0000123456

@

xxx.xxx.xxx.xxx

>;tag=as04bc6a95

To: <sip: [e-mail]>
Call-ID: [email protected]
CSeq: 1749 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120

Contact: <sip: 0000123456@

000.000.000.000

>

Event: registration
Content-Length: 0

4.1.2 GUEST → PBX

SIP/2.0 100 Trying

Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;rport=5060
From: <sip: [e-mail]>;tag=as04bc6a95
To: <sip: [e-mail]>
Call-ID: [email protected]
CSeq: 1749 REGISTER
User-Agent: Asterisk PBX

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

Supported: replaces
Contact: <sip: [e-mail]>
Content-Length: 0

4

.1.3 GUEST →

PBX

4

.1.3 GUEST →

PBX

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;rport=5060
From: <sip: [e-mail]>;tag=as04bc6a95
To: <sip: [e-mail]>;tag=as245298a3
Call-ID: [email protected]
CSeq: 1749 REGISTER

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx", nonce="3deff552"
Content-Length: 0

27

４．Technical Data

4.1.4 PBX → GUEST

REGISTER sip: xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;rport

From: <sip:

0000123456

@

xxx.xxx.xxx.xxx

>;tag=as2031f6e2

To: <sip: [e-mail] >
Call-ID: [email protected] CSeq: 1750 REGISTER
User-Agent: Asterisk PBX Max-Forwards: 70
Authorization: Digest username="0000123456", realm=" xxx.xxx.xxx.xxx ", algorithm=MD5,
uri="sip: xxx.xxx.xxx.xxx", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056", opaque=""
Expires: 120

Contact: <sip:

0000123456

@

000.000.000.000

> Event: registration

Content-Length: 0

4.1.5 GUEST → PBX

SIP/2.0 100 Trying

Via:SIP

/2.0/UDP

000.000.000.000

:5060;branch=z9hG4bK1db71efa;received=

000.000.000.000

;rport=5060

From: <sip: [e-mail] >;tag=as2031f6e2
To: <sip: [e-mail] >
Call-ID: [email protected] CSeq: 1750 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces

Contact: <sip:

0000123456

@

xxx.xxx.xxx.xxx

>

Content-Length: 0

4.1.6 GUEST → PBX

SIP/2.0 200 OK

SIP/2.0 200 OK

Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;received=000.000.000.000;rport=5060
From: <sip: [e-mail] >;tag=as2031f6e2
To: <sip: [e-mail] >;tag=as245298a3
Call-ID: [email protected]
CSeq: 1750 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Expires: 120
Contact: <sip: 0000123456@000.000.000.000>;expires=120
Date: Mon, 05 Jul 2010 04:20:13 GMT
Content-Length: 0

28

４．Technical Data

4.2. SIP INVITE message of outgoing call from your IP-PBX through SIP Trunk 2

SIP From header should be :

From: “Phone Display name”<sip:CallerID@SIP Trunk 2 IP address or FQDN＞

INVITE
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee

To: <sip:[e-mail]>

SIP Trunk 2

xxx.xxx.xxx.xxx

your IP-PBX

000.000.000.000

Phone

Display Name

CallerID

IP address of SIP

Trunk 2 server

Call-ID: 6426c31c421e503b72515b46569f2ee0@

xxx.xxx.xxx.xxx

407 Proxy Authentication Required
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as4abe0e65
Call-ID: [e-mail]

ACK
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as4abe0e65
Call-ID: [e-mail]

1

2

Receiver

Phone

Number

INVITE(with credential information)
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>
Call-ID: [e-mail]

100 Trying
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>
Call-ID: [e-mail]

3

4

180 Ringing
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as54380085
Call-ID: [e-mail]

183 Session Progress
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as54380085
Call-ID: [e-mail]

5

6

200 OK
From: "aiueo PBX" <sip

0312123434

@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as54380085
Call-ID: [e-mail]

ACK
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee

To: <sip:[e-mail]>;tag=as54380085

starting a call

7

8

Call-ID: 6426c31c421e503b72515b46569f2ee0@

xxx.xxx.xxx.xxx

BYE
From: <sip:[e-mail]>;tag=as54380085
To: "aiueo PBX" <sip

0312123434

@000.000.000.000>;tag=as5dd4eaee

Call-ID: [e-mail]

200 OK

Terminating a call

9

10

From: <

sip:080AAAAXXXX@

xxx.xxx.xxx.xxx

>;tag=as54380085

To: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as5dd4eaee

Call-ID: [e-mail]

11

４．Technical Data

4.2.1 PBX → GUEST

INVITE sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rport

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as5dd4eaee

To: <sip:[e-mail]>
Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]

CSeq: 102 INVITE
User-Agent: Asterisk PBX Max-Forwards: 70
Date: Fri, 02 Jul 2010 03:05:26 GMT

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

Content-Type: application/sdp Content-Length: 267

v=0
o=root 22702 22702 IN IP4 000.000.000.000
s=session
c=IN IP4 000.000.000.000
t=0 0

m=audio 18572 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

4.2.2 GUEST → PBX

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;received=000.000.000.000;rport=5060

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as5dd4eaee

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as5dd4eaee

To: <sip:[e-mail]>;tag=as4abe0e65
Call-ID: [e-mail]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="23a44cfd"

Content

-

Length: 0

30

４．Technical Data

4.2.3 PBX → GUEST

ACK sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rport

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as5dd4eaee

To: <sip:[e-mail]>;tag=as4abe0e65
Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70

Content

-

Length: 0

4.2.4 PBX → GUEST

INVITE sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;rport
From: "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee

To: <sip:080AAAAXXXX@

xxx.xxx.xxx.xxx

>

Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX Max-Forwards: 70
Proxy-Authorization: Digest username=" 0000123456 ", realm="xxx.xxx.xxx.xxx ",
algorithm=MD5, uri="sip:[e-mail]", nonce="23a44cfd",

response="cc6c5a668cbd435dee31c767981ff710", opaque=""

Date: Fri, 02 Jul 2010 03:05:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp Content-Length: 267

v=0
o=root 22702 22703 IN IP4 000.000.000.000

s=session

s=session

c=IN IP4 000.000.000.000
t=0 0
m=audio 18572 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000

a=fmtp:101 0

-

16

a=silenceSupp:off - - - -

31

４．Technical Data

4.2.5 GUEST → PBX

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as5dd4eaee

To: <sip:[e-mail]>
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces

Contact: <sip:080AAAAXXXX@

xxx.xxx.xxx.xxx

>

Content-Length: 0

4.2.6. GUEST → PBX

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as5dd4eaee

To: <sip:[e-mail]>;tag=as54380085
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces

Contact: <sip:080AAAAXXXX@

xxx.xxx.xxx.xxx

>

Content-Length: 0

32

４．Technical Data

4.2.7 GUEST → PBX

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP

000.000.000.000

:5060;branch=z9hG4bK4fc267d7;received=

000.000.000.000

;rport=5060

From: "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as54380085
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:[e-mail]>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx
s=session

c=IN IP4

xxx.xxx.xxx.xxx

t=0 0
m=audio 18922 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

33

４．Technical Data

4.2.8 GUEST → PBX

SIP/2.0 200 OK

Via: SIP/2.0/UDP

000.000.000.000

:5060;branch=z9hG4bK4fc267d7;received=

000.000.000.000

;rport=5060

From: "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as54380085
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:[e-mail]>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 4414 4415 IN IP4 xxx.xxx.xxx.xxx
s=session

c=IN IP4

xxx.xxx.xxx.xxx

t=0 0
m=audio 18922 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

4.2.9 PBX → GUEST

ACK sip:[e-mail] SIP/2.0

Via: SIP/2.0/UDP

000.000.000.000

:5060;branch=z9hG4bK6c101c7f;rport

Via: SIP/2.0/UDP

000.000.000.000

:5060;branch=z9hG4bK6c101c7f;rport

From: " aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as5dd4eaee
To: <sip:[e-mail]>;tag=as54380085
Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70

Content

-

Length: 0

34

４．Technical Data

4.2.10 GUEST → PBX

BYE sip:0312123434@000.000.000.000 SIP/2.0

Via: SIP/2.0/UDP

xxx.xxx.xxx.xxx

:5060;branch=z9hG4bK166bf514;rport

From: <sip:[e-mail]>;tag=as54380085
To: "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee
Call-ID: [e-mail]
CSeq: 102 BYE
User-Agent: Asterisk PBX Max-Forwards: 70
Content-Length: 0

4.2.11. PBX → GUEST

SIP/2.0 200 OK Via:SIP/2.0/UDP

xxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;received=xxx.xxx.xxx.xxx;rport=5060

From: <sip:[e-mail]>;tag=as54380085
To: " aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as5dd4eaee

Call-ID: 6426c31c421e503b72515b46569f2ee0@

xxx.xxx.xxx.xxx

CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0312123434@000.000.000.000>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

35

４．Technical Data

4.3. SIP Busy message while outgoing call in case receiver is on another call

Busy message sent by SIP Trunk 2 when receiver is currently on another call,

SIP Trunk 2

xxx.xxx.xxx.xxx

your IP-PBX

000.000.000.000

CallerID

IP address of SIP Trunk 2

server

INVITE
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as48ac6d56

1

2

To: <

sip:

080AAAAXXXX

@

xxx.xxx.xxx.xxx

>

Call-ID: [e-mail]

407 Proxy Authentication Required
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as48ac6d56

To: <sip:[e-mail]>;tag=as291aca90
Call-ID: [e-mail]

3

ACK
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as48ac6d56

To: <sip:[e-mail]>;tag=as291aca90
Call-ID: [e-mail]

INVITE(with authentication information)

From: "

aiueo

PBX

" <

sip:

0312123434

@

000.000.000.000

>

;tag=as48ac6d56

4

5

From: "

aiueo

PBX

" <

sip:

0312123434

@

000.000.000.000

>

;tag=as48ac6d56

To: <sip:[e-mail]>
Call-ID: [e-mail]

100 Trying
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as48ac6d56

To: <sip:[e-mail]>
Call-ID: [e-mail]

5

6

SIP/2.0 486 Busy Here
From: "aiueo PBX" <sip:

0312123434

@000.000.000.000>;tag=as48ac6d56

To: <sip:[e-mail]>;tag=as715c3c5e
Call-ID: [e-mail]

ACK
From: "aiueo PBX" <sip:

3120123434

@000.000.000.000>;tag=as48ac6d56

7

To: <sip:[e-mail]>;tag=as715c3c5e
Call-ID: [e-mail]

36

figure 4.3 SIP flow including Busy message while outgoing call

４．Technical Data

4.3.1 PBX → GUEST

INVITE sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rport

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as48ac6d56

To: <sip:[e-mail]>
Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]
CSeq: 102 INVITE
User-Agent: Asterisk PBX Max-Forwards: 70
Date: Tue, 06 Jul 2010 10:09:37 GMT

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

Content-Type: application/sdp Content-Length: 267

v=0
o=root 22702 22702 IN IP4 000.000.000.000
s=session
c=IN IP4 000.000.000.000
t=0 0

m=audio 14646 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

4.3.2 GUEST→ PBX

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;received=000.000.000.000;rport=5060
From: " aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as48ac6d56

To: <sip:

080AAAAXXXX

@

xxx.xxx.xxx.xxx

>;tag=as291aca90

To: <sip:

080AAAAXXXX

@

xxx.xxx.xxx.xxx

>;tag=as291aca90

Call-ID: [e-mail]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="15a6e863"
Content-Length: 0

37

４．Technical Data

4.3.3 PBX → GUEST

ACK sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rport

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as48ac6d56

To: <sip:[e-mail] >;tag=as291aca90
Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70

Content

-

Length: 0

4.3.4 PBX→GUEST

INVITE sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rport
From: " aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as48ac6d56

To: <sip:

080AAAAXXXX

@xxx.xxx.xxx.xxx>

Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX Max-Forwards: 70
Proxy-Authorization: Digest username="0000123456", realm="xxx.xxx.xxx.xxx ",
algorithm=MD5, uri="sip:[e-mail] ", nonce="15a6e863",

response="54ebd3bdb5bab4b621f55fbd3ffe5e0b", opaque=""

Date: Tue, 06 Jul 2010 10:09:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0

o=root 22702 22703 IN IP4

000.000.000.000

o=root 22702 22703 IN IP4

000.000.000.000

s=session
c=IN IP4 000.000.000.000
t=0 0
m=audio 14646 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone

-

event/8000

a=fmtp:101 0-16
a=silenceSupp:off - - - -

38

４．Technical Data

4.3.5 GUEST→ PBX

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as48ac6d56

To: <sip:[e-mail]>
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces

Contact: <sip:

0312123434

@

xxx.xxx.xxx.xxx

>

Content-Length: 0

4.3.6. GUEST → PBX

SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060

From: "

aiueo

PBX

" <sip:

0312123434

@

000.000.000.000

>;tag=as48ac6d56

To: <sip:[e-mail]>;tag=as715c3c5e
Call-ID: [e-mail]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Contact: <sip:[e-mail]>
Content-Length: 0

4.3.7 PBX → GUEST

ACK sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rport
From: " aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as48ac6d56

To: <sip:

080AAAAXXXX

@

xxx.xxx.xxx.xxx

>;tag=as715c3c5e

To: <sip:

080AAAAXXXX

@

xxx.xxx.xxx.xxx

>;tag=as715c3c5e

Contact: <sip:0312123434@000.000.000.000>
Call-ID: [e-mail]
CSeq: 103 ACK
User-Agent: Asterisk PBX Max-Forwards: 70
Content-Length: 0

39

４．Technical Data

4.4. SIP INVITE message of incoming call from SIP Trunk 2 to your IP-PBX

SIP To header will be :

To: <sip:Recipient Phone Number@Your IP PBX IP address＞

*SIP Trunk 2 sets the same recipient phone number to Alert

-

info header as well.

SIP Trunk 2

xxx.xxx.xxx.xxx

your IP-PBX

000.000.000.000

IP address of

CallerID

INVITE
From: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a

Recipient

your IP

-

PBX

1

2

To: <

sip:

0312345678

@

000.000.000.000

>

Call-ID: [e-mail]

100 Trying
From: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a

To: <sip:

0312345678

@000.000.000.000>

Call-ID: [e-mail]

IP address of SIP

Trunk 2 server

3

200 OK
From: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a
To: <sip:

0312345678

@000.000.000.000>;tag=as577af7ce

Call-ID: [e-mail]

ACK
From: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a

Starting a call

4

5

To: <

sip:

0312345678

@

000.000.000.000

>;tag=as577af7ce

Call-ID: [e-mail]

BYE
From: <sip:

0312345678

@000.000.000.000>;tag=as577af7ce
To: “ 080AAAAXXXX " <sip:[e-mail]>;tag=as1dddca7a
Call-ID: [e-mail]

Terminating a call

5

6

200 OK
From: <sip:

0312345678

@000.000.000.000>;tag=as577af7ce
To: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a
Call-ID: [e-mail]

40

figure 4.4 SIP INVITE flow (incoming)

４．Technical Data

4.4.1 GUEST→PBX

INVITE sip:0312345678@000.000.000.000 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;rport

From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a

To: <sip: 0312345678 @000.000.000.000>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX Max-Forwards: 70
Date: Fri, 02 Jul 2010 05:41:33 GMT

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

Supported: replaces
X-Asterisk-Guest-Tag: 00008
X-Asterisk-Guest-Uniqueid: 1278049293.36
Alert-info: 0312345678
Content-Type: application/sdp
Content-Length: 242

v=0

o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 15224 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone

-

event/8000

a=fmtp:101 0-16
a=silenceSupp:off - - - ­a=ptime:20
a=sendrecv

4.4.2. GUEST←PBX

4.4.2. GUEST←PBX

SIP/2.0 100 Trying
Via:SIP/2.0/UDP

xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;received=xxx.xxx.xxx.xxx;rport=5060

From: "080AAAAXXXX" <sip: 080AAAAXXXX @xxx.xxx.xxx.xxx>;tag=as1dddca7a
To: <sip:0312345678@000.000.000.000>
Call-ID: [e-mail]

CSeq

: 102 INVITE

User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0312345678@000.000.000.000>
Content-Length: 0

41

４．Technical Data

4.4.3. GUEST ←PBX

SIP/2.0 200 OK
Via:SIP/2.0/UDP

xxx.xxx.xxx.xxx

:5060;branch=z9hG4bK546a1def;received=

xxx.xxx.xxx.xxx

;rport

=5060

From: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a
To: <sip:0312345678@000.000.000.000>;tag=as577af7ce
Call-ID: [e-mail]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Contact: <sip:

0312345678

@

000.000.000.000

>

Content-Type: application/sdp
Content-Length: 220

v=0
o=root 22702 22702 IN IP4 000.000.000.000
s=session
c=IN IP4 000.000.000.000

t=0 0

m=audio 18182 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

4.4.4 GUEST →PBX

ACK sip:0312345678@000.000.000.000 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3afc8626;rport
From: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a
To: <sip:0312345678@000.000.000.000>;tag=as577af7ce

Contact: <sip:080AAAAXXXX@

xxx.xxx.xxx.xxx

>

Contact: <sip:080AAAAXXXX@

xxx.xxx.xxx.xxx

>

Call-ID: [e-mail]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

42

４．Technical Data

4.4.5. GUEST ←PBX

BYE sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;rport

From: <sip:

0312345678

@

000.000.000.000

>;tag=as577af7ce

To: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a
Call-ID: [e-mail]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

4.4.6. GUEST →PBX

SIP/2.0 200 OK
Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;received=000.000.000.000;rport=5060
From: <sip:0312345678@000.000.000.000>;tag=as577af7ce
To: "080AAAAXXXX" <sip:[e-mail]>;tag=as1dddca7a

Call-ID: 490e49cf2141339f0007e5ce47d80dd1@

xxx.xxx.xxx.xxx

CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[e-mail]>
Content-Length: 0

43

４．Technical Data

4.5. SIP Busy message while incoming call in case receiver is on another call

Busy message sent by SIP Trunk 2 when receiver is currently on another call,

SIP Trunk 2

xxx.xxx.xxx.xxx

your IP-PBX

000.000.000.000

IP address of

CallerID

INVITE

From:

"

080AAAAXXXX

" <sip:

080AAAAXXXX"

@

xxx.xxx.xxx.xxx

>;tag=as0f1a5f0c

Recipient

IP address of

your IP-PBX

SIP Trunk 2

server

1

2

From:

"

080AAAAXXXX

" <sip:

080AAAAXXXX"

@

xxx.xxx.xxx.xxx

>;tag=as0f1a5f0c

To: <sip:

0312345678

@000.000.000.000>

Call-ID: [e-mail]

100 Trying
From: "080AAAAXXXX" <sip:[e-mail]>;tag=as0f1a5f0c
To: <sip:

0312345678

@000.000.000.000>

Call-ID: [e-mail]

3

486 Busy Here
From: "080AAAAXXXX" <sip:[e-mail]>;tag=as0f1a5f0c
To: <sip:

0312345678

@000.000.000.000>

Call-ID: [e-mail]

ACK

4

From:

"

080AAAAXXXX

" " <sip:

080AAAAXXXX

@

xxx.xxx.xxx.xxx

>;tag=as0f1a5f0c

To: <sip:

0312345678

@000.000.000.000>

Call-ID: 6dd7b12f1438e1572cae057f274419e6@000.000.000.000

figure 4.5 SIP flow including Busy message while incoming call

44

４．Technical Data

4.5.1 GUEST → PBX

INVITE sip:0312345678@000.000.000.000 SIP/2.0
Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;rport

From:"

080AAAAXXXX

"<sip:0311112222@

xxx.xxx.xxx.xxx

>;tag=as0f1a5f0c

To: <sip:0312345678@000.000.000.000>
Contact: <sip: [e-mail]>
Call-ID: [e-mail]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70

Date: Fri, 09 Jul 2010 02:27:46 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-Asterisk-Guest-Tag: 00024
X-Asterisk-Guest-Uniqueid: 1278642466.508
Alert-info: 0312345678
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 10408 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - ­a=ptime:20
a=sendrecv

4.5.2 PBX → GUEST

SIP/2.0 100 Trying Via: SIP/2.0/UDP

xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;received=xxx.xxx.xxx.xxx;rport=5060

From: "080AAAAXXXX" <sip:[e-mail]>;tag=as0f1a5f0c
To: <sip:0312345678@000.000.000.000>

Call-ID: 1aa4d60711e0817d731834f474d958b0@

xxx.xxx.xxx.xxx

CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0312345678@000.000.000.000>
Content-Length: 0

45

４．Technical Data

4.5.3. PBX → GUEST

SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP

xxx.xxx.xxx.xxx

:5060;branch=z9hG4bK0b7fb7b8;received=

xxx.xxx.xxx.xxx

;rport

=5060

From: " 080AAAAXXXX" <sip:[e-mail]>;tag=as0f1a5f0c
To: <sip:0312345678@000.000.000.000>
Call-ID: [e-mail]
CSeq: 102 INVITE
Contact: <sip:0312345678@000.000.000.000>
Content-Length: 0

4.5.4. GUEST→ PBX

Transmitting (NAT) to GUEST
ACK sip: [e-mail] SIP/2.0
Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;rport

From:"

080AAAAXXXX

"<sip:

080AAAAXXXX

@

xxx.xxx.xxx.xxx

>;tag=as0f1a5f0c

To: <sip:0312345678@000.000.000.000>
Contact: <sip:080AAAAXXXX@000.000.000.000>
Call-ID: 6dd7b12f1438e1572cae057f274419e6@000.000.000.000
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70

Content

-

Length: 0

46