SIP Trunk 2 IP-PBX User Guide Asterisk - agileSIP Trunk 2 is a next generation IP phone service that...

SIP Trunk 2 IP-PBX User Guide（Asterisk）

Ver1.0.0 2015/08/01Ver1.0.3 2015/09/17Ver1.0.4 2015/10/07Ver1.0.5 2015/10/15Ver1.0.6 2015/10/23Ver1.0.7 2016/01/18

2

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

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.

＜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.xAsterisk PBX 1.6.xAsterisk PBX 1.8.xAsterisk PBX 11Asterisk PBX 12

・Aspire XIP3WW-32VOIPDB-A1version: 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.

Our Server IP address list *as of Oct 23, 2015221.243.8.194221.243.8.195101.110.51.82101.110.51.83113.41.163.2113.41.163.3

１．SIP Trunk 2 Overview

3

Ext. 200 Ext. 201

4

１．SIP Trunk 2 Overview

To:<sip:[email protected]>

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

From: <sip:[email protected]>

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

Image 1. Configuration Diagram of Incoming/Outgoing Calls

xxx.xxx.xxx.xxxSIP Trunk 2

Your IP-PBX

DID: 0312123434DID: 0312345678

0000.0000.0000.0000

*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******]

5

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 Select “Purchase” at the top menu and choose ”Purchase Unique”in Circle Management Page

② Select quantity of SIP trunk 2③ Click “Add to Cart” to proceed for your purchase

③

①

②

6


Purchase phone number here*At least one phone number will be needed for external phone calls through SIP Trunk＜Phone Number Purchase Screen>

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.

①

②

7


＜SIP Trunk 2 List＞

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

①② ③

0000123456

8


＜SIP Trunk 2 Detailed Settings ・ Password Authentication＞

① Login server name of SIP Trunk 2② Unique is used as client user ID of your user PBX end.③ Item “Name” is where you can name/rename your SIP Trunk account.④ Select your desired authentication method from

“Password Authentication” or “Authentication with IP Address” or“Authentication using both IP Address and Password”

⑤ Enter your terminal password is used as client user password of your PBX end. ⑥ 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.

xxx.xxx.xxx.xxx①②③④⑤⑥

0000123456

9


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



⑤ Enter a public IP address / a port number of your IP-PBX *You can add multiple IP addresses/ports from “+Insert” button.⑥ Your IP-PBX will receive incoming call if ticked. *If unticked it will work only for outgoing calls.⑦ Set multiple call count. It’s 1 by default. Purchase “Additional 1 channel for


xxx.xxx.xxx.xxx①②③④

⑤ ⑥

⑦

0000123456

10


＜SIP Trunk 2 Detailed Settings ・Authentication using both IP Address and Password>



⑤ Enter your terminal password is used as client user password of your PBX end. ⑥ Enter a public IP address of your IP-PBX.⑦ Set multiple call count. It’s 1 by default. Purchase “Additional 1 channel for


①②③④⑤⑥⑦

xxx.xxx.xxx.xxx

0000123456

11


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

＜Phone Number List>

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

②

①

〔0000123456〕

12

３．Configuration Example of your IP-PBX

3.1. Configuration Example in Asterisk

[Account Example]Unique: 0000123456Password: passwordDIDs: 0312345678 , 0312123434Extensions: 200, 201Login 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: 0312345678Outgoing 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=3600port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulawlanguage=jp

register => 0000123456:password@siptr

[siptr]type=friendusername=0000123456 secret=password context=inbound canreinvite=no host=xxx.xxx.xxx.xxxinsecure=port,invitedisallow=allallow=ulawqualify=yesnat=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>

13



[200]type=friend username=200secret=200pass host=dynamic context=outbound-1


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

14


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

[general]allowguest=nomaxexpirey=3600defaultexpirey=3600port=5060bindaddr=0.0.0.0srvlookup=yesdisallow=allallow=ulawlanguage=jp

[siptr]type=friendcontext=inboundcanreinvite=nohost=xxx.xxx.xxx.xxxinsecure=port,invitedisallow=allallow=ulawqualify=yesnat=yes

[peer1]type=friendcontext=inboundhost=221.243.8.194nat=yes





15


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



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



16


; ------------------; 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)= 0312345678exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T)exten => _0., 3,Congestionexten => _0.,104,Busy

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

17


[outbound-2]exten => _0., 1,Set(CALLERID(num)= 0312123434)exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T)exten => _0., 3,Congestionexten => _0.,104,Busy

exten => _1., 1,Set(CALLERID(num)= 0312123434)exten => _1., 2,Dial(SIP/${EXTEN}@siptr,120,T)exten => _1., 3,Congestionexten => _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,Congestionexten => _ XXX, 102,Busy; XXX represents 3 digit-extensions. Please adjust digit number as yours.

18


Group １:Max multiple count 2Extensions 201 ～ 202Phone Numbers 03-1234-5678

Group ２:Max multiple count 3Extensions 301 ～ 302Phone Numbers 03-1212-3434

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

[Settings Example]Set max multiple call count (for external calls) as 2 for Group 1Set max multiple call count (for external calls) as 3 for Group 2


[general]allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extdport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulawlanguage=jp

register=>0000123456:[email protected]/0000123456

[0000123456]type=friendusername=0000123456secret=password host=xxx.xxx.xxx.xxxinsecure=port,invitecontext=inboundqualify=yesnat=yes;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11


19



; Group 1[201]type=friend context=group1_outbound username=201secret=password host=dynamic

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



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

20

３．Configuration Example of your IP-PBX;--------------;sip.conf (IP address authentication) ;--------------

[general]allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extdport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulawlanguage=jp

[siptr]type=friendcontext=inboundcanreinvite=nohost= xxx.xxx.xxx.xxxinsecure=port,invitedisallow=allallow=ulawqualify=yesnat=yes

[peer1]type=friend context=inboundhost=221.243.8.194nat=yes





21


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



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





22


<extensions.conf Example in your Asterisk>

; ------------------; 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,Congestionexten => 0312345678,106,Busy

; Group 2exten => 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

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

23


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

exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T)exten => _ XXX, 2,Congestionexten => _ 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,Congestionexten => _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,Congestionexten => _1.,106,Busy

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

24

４．Technical Data

4.1. SIP REGISTER message：

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

figure 4.1 SIP flow for REGISTER

※Sending REGISTER message is NOT required in case your authentication method is “Authentication with IP Address”

REGISTERFrom: <sip: [email protected]>;tag=as04bc6a95To: <sip: [email protected]>Call-ID: [email protected]

your IP-PBX

000.000.000.000SIP Trunk 2

xxx.xxx.xxx.xxx

1100 TryingFrom: <sip: [email protected]>;tag=as04bc6a95To: <sip: [email protected]>Call-ID: [email protected]

2401 UnauthorizedFrom: <sip: [email protected]>;tag=as04bc6a95To: <sip: [email protected]>;tag=as245298a3Call-ID: [email protected]

3REGISTER(with credential information)From: <sip: [email protected]>;tag=as2031f6e2To: <sip: [email protected]>Call-ID: [email protected]

4SIP/2.0 100 TryingFrom: <sip: [email protected]>;tag=as2031f6e2To: <sip: [email protected]>Call-ID: [email protected]

5200 OKFrom: <sip: [email protected]>;tag=as2031f6e2To: <sip: [email protected]>;tag=as245298a3Call-ID: [email protected]

6

Your ID (SIP Trunk 2 unique number

IP address of SIP Trunk 2

25


4.1.1 PBX → GUEST

REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;rport From: <sip: [email protected]>;tag=as04bc6a95To: <sip: [email protected]>Call-ID: [email protected]: 1749 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip: [email protected]> Event: registrationContent-Length: 0

4.1.2 GUEST → PBX

SIP/2.0 100 TryingVia:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;rport=5060 From: <sip: [email protected]>;tag=as04bc6a95To: <sip: [email protected]>Call-ID: [email protected] CSeq: 1749 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip: [email protected]>Content-Length: 0

4.1.3 GUEST → PBX

SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;rport=5060 From: <sip: [email protected]>;tag=as04bc6a95To: <sip: [email protected]>;tag=as245298a3 Call-ID: [email protected]: 1749 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesWWW-Authenticate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx", nonce="3deff552" Content-Length: 0

26


4.1.4 PBX → GUEST

REGISTER sip: xxx.xxx.xxx.xxx SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;rport From: <sip: [email protected] >;tag=as2031f6e2To: <sip: [email protected] >Call-ID: [email protected] CSeq: 1750 REGISTERUser-Agent: Asterisk PBX Max-Forwards: 70Authorization: Digest username="0000123456", realm=" xxx.xxx.xxx.xxx ", algorithm=MD5, uri="sip: xxx.xxx.xxx.xxx", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056", opaque=""Expires: 120Contact: <sip: [email protected]> Event: registrationContent-Length: 0

4.1.5 GUEST → PBX

SIP/2.0 100 TryingVia:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;received=000.000.000.000;rport=5060 From: <sip: [email protected] >;tag=as2031f6e2To: <sip: [email protected] >Call-ID: [email protected] CSeq: 1750 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip: [email protected] >Content-Length: 0

4.1.6 GUEST → PBX

SIP/2.0 200 OKVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;received=000.000.000.000;rport=5060 From: <sip: [email protected] >;tag=as2031f6e2To: <sip: [email protected] >;tag=as245298a3 Call-ID: [email protected]: 1750 REGISTERUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesExpires: 120Contact: <sip: [email protected]>;expires=120 Date: Mon, 05 Jul 2010 04:20:13 GMTContent-Length: 0


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＞

INVITEFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>Call-ID: [email protected]

407 Proxy Authentication RequiredFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65Call-ID: [email protected]

ACKFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65Call-ID: [email protected]

INVITE(with credential information)From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>Call-ID: [email protected]

100 TryingFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>Call-ID: [email protected]

180 RingingFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085Call-ID: [email protected]

183 Session ProgressFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085Call-ID: [email protected]

200 OKFrom: "aiueo PBX" <[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085Call-ID: [email protected]

ACKFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085Call-ID: [email protected]

BYEFrom: <sip:[email protected]>;tag=as54380085To: "aiueo PBX" <[email protected]>;tag=as5dd4eaee Call-ID: [email protected]

200 OKFrom: <sip:[email protected]>;tag=as54380085To: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee Call-ID: [email protected]

SIP Trunk 2xxx.xxx.xxx.xxx

your IP-PBX000.000.000.000

Phone Display Name CallerID

IP address of SIP Trunk 2 server

starting a call

Terminating a call

1

2

3

4

5

6

7

8

9

10

11

Receiver Phone

Number

28


4.2.1 PBX → GUEST

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rportFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>Contact: <sip:[email protected]>Call-ID: [email protected]: 102 INVITEUser-Agent: Asterisk PBX Max-Forwards: 70Date: Fri, 02 Jul 2010 03:05:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdp Content-Length: 267

v=0o=root 22702 22702 IN IP4 000.000.000.000s=sessionc=IN IP4 000.000.000.000t=0 0m=audio 18572 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -

4.2.2 GUEST → PBX

SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaeeTo: <sip:[email protected]>;tag=as4abe0e65Call-ID: [email protected] CSeq: 102 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="23a44cfd" Content-Length: 0

29


4.2.3 PBX → GUEST

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rportFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65Contact: <sip:[email protected]>Call-ID: [email protected]: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0

4.2.4 PBX → GUEST

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;rportFrom: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>Contact: <sip:[email protected]>Call-ID: [email protected]: 103 INVITEUser-Agent: Asterisk PBX Max-Forwards: 70Proxy-Authorization: Digest username=" 0000123456 ", realm="xxx.xxx.xxx.xxx ",algorithm=MD5, uri="sip:[email protected]", nonce="23a44cfd", response="cc6c5a668cbd435dee31c767981ff710", opaque=""Date: Fri, 02 Jul 2010 03:05:26 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdp Content-Length: 267


30


4.2.5 GUEST → PBX

SIP/2.0 100 TryingVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaeeTo: <sip:[email protected]>Call-ID: [email protected]: 103 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:[email protected]>Content-Length: 0

4.2.6. GUEST → PBX

SIP/2.0 180 RingingVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaeeTo: <sip:[email protected]>;tag=as54380085Call-ID: [email protected]: 103 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:[email protected]>Content-Length: 0

31


4.2.7 GUEST → PBX

SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaeeTo: <sip:[email protected]>;tag=as54380085Call-ID: [email protected]: 103 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 242

v=0o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxxs=sessionc=IN IP4 xxx.xxx.xxx.xxxt=0 0m=audio 18922 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv

32


4.2.8 GUEST → PBX

SIP/2.0 200 OKVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaeeTo: <sip:[email protected]>;tag=as54380085Call-ID: [email protected]: 103 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:[email protected]>Content-Type: application/sdpContent-Length: 242

v=0o=root 4414 4415 IN IP4 xxx.xxx.xxx.xxxs=sessionc=IN IP4 xxx.xxx.xxx.xxxt=0 0m=audio 18922 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv

4.2.9 PBX → GUEST

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK6c101c7f;rportFrom: " aiueo PBX " <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085Contact: <sip:[email protected]>Call-ID: [email protected]: 103 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0

33


4.2.10 GUEST → PBX

BYE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;rport From: <sip:[email protected]>;tag=as54380085To: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee Call-ID: [email protected]: 102 BYEUser-Agent: Asterisk PBX Max-Forwards: 70Content-Length: 0

4.2.11. PBX → GUEST

SIP/2.0 200 OK Via:SIP/2.0/UDPxxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;received=xxx.xxx.xxx.xxx;rport=5060 From: <sip:[email protected]>;tag=as54380085To: " aiueo PBX " <sip:[email protected]>;tag=as5dd4eaee Call-ID: [email protected]: 102 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:[email protected]> Content-Length: 0X-Asterisk-HangupCause: Normal Clearing

34


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,

figure 4.3 SIP flow including Busy message while outgoing call

SIP Trunk 2 xxx.xxx.xxx.xxx

your IP-PBX000.000.000.000 CallerID


1

2

3

4

5

6

7

INVITEFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>Call-ID: [email protected]

407 Proxy Authentication RequiredFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>;tag=as291aca90Call-ID: [email protected]

ACKFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>;tag=as291aca90Call-ID: [email protected]

INVITE(with authentication information)From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>Call-ID: [email protected]

100 TryingFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>Call-ID: [email protected]

SIP/2.0 486 Busy HereFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>;tag=as715c3c5eCall-ID: [email protected]

ACKFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>;tag=as715c3c5eCall-ID: [email protected]

35


4.3.1 PBX → GUEST

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rportFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>Contact: <sip:[email protected]>Call-ID: [email protected]: 102 INVITEUser-Agent: Asterisk PBX Max-Forwards: 70Date: Tue, 06 Jul 2010 10:09:37 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdp Content-Length: 267

v=0o=root 22702 22702 IN IP4 000.000.000.000s=sessionc=IN IP4 000.000.000.000t=0 0m=audio 14646 RTP/AVP 0 8 3 101a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16a=silenceSupp:off - - - -

4.3.2 GUEST→ PBX

SIP/2.0 407 Proxy Authentication RequiredVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>;tag=as291aca90Call-ID: [email protected]: 102 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesProxy-Authenticate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="15a6e863" Content-Length: 0

36


4.3.3 PBX → GUEST

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rportFrom: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected] >;tag=as291aca90Contact: <sip:[email protected]>Call-ID: [email protected] CSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0

4.3.4 PBX→GUEST

INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rportFrom: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>Contact: <sip:[email protected]>Call-ID: [email protected] CSeq: 103 INVITEUser-Agent: Asterisk PBX Max-Forwards: 70Proxy-Authorization: Digest username="0000123456", realm="xxx.xxx.xxx.xxx ",algorithm=MD5, uri="sip:[email protected] ", nonce="15a6e863", response="54ebd3bdb5bab4b621f55fbd3ffe5e0b", opaque=""Date: Tue, 06 Jul 2010 10:09:37 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Type: application/sdpContent-Length: 267


37


4.3.5 GUEST→ PBX

SIP/2.0 100 TryingVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>Call-ID: [email protected]: 103 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:[email protected]>Content-Length: 0

4.3.6. GUEST → PBX

SIP/2.0 486 Busy HereVia: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56To: <sip:[email protected]>;tag=as715c3c5eCall-ID: [email protected]: 103 INVITEUser-Agent: Asterisk PBXContact: <sip:[email protected]>Content-Length: 0

4.3.7 PBX → GUESTACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rportFrom: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5eContact: <sip:[email protected]>Call-ID: [email protected]: 103 ACKUser-Agent: Asterisk PBX Max-Forwards: 70Content-Length: 0

38


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.

figure 4.4 SIP INVITE flow (incoming)


your IP-PBX000.000.000.000

IP address of your IP-PBX

1

2

3

4

5

6

CallerID

INVITEFrom: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7aTo: <sip:[email protected]>Call-ID: [email protected]

100 TryingFrom: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7aTo: <sip:[email protected]>Call-ID: [email protected]

200 OKFrom: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7aTo: <sip:[email protected]>;tag=as577af7ceCall-ID: [email protected]

ACKFrom: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7aTo: <sip:[email protected]>;tag=as577af7ceCall-ID: [email protected]

BYEFrom: <sip:[email protected]>;tag=as577af7ceTo: “ 080AAAAXXXX " <sip:[email protected]>;tag=as1dddca7aCall-ID: [email protected]

200 OKFrom: <sip:[email protected]>;tag=as577af7ceTo: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7aCall-ID: [email protected]

Recipient


Starting a call

Terminating a call

39


4.4.1 GUEST→PBX

INVITE sip:[email protected] SIP/2.0Via: 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 INVITEUser-Agent: Asterisk PBX Max-Forwards: 70Date: Fri, 02 Jul 2010 05:41:33 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesX-Asterisk-Guest-Tag: 00008X-Asterisk-Guest-Uniqueid: 1278049293.36Alert-info: 0312345678 Content-Type: application/sdpContent-Length: 242

v=0o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxxs=sessionc=IN IP4 xxx.xxx.xxx.xxxt=0 0m=audio 15224 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv

4.4.2. GUEST←PBX

SIP/2.0 100 Trying Via:SIP/2.0/UDPxxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip: 080AAAAXXXX @xxx.xxx.xxx.xxx>;tag=as1dddca7aTo: <sip:[email protected]>Call-ID: [email protected]: 102 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:[email protected]> Content-Length: 0

40


4.4.3. GUEST ←PBX

SIP/2.0 200 OK Via:SIP/2.0/UDPxxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7aTo: <sip:[email protected]>;tag=as577af7ceCall-ID: [email protected]: 102 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:[email protected]> Content-Type: application/sdpContent-Length: 220

v=0o=root 22702 22702 IN IP4 000.000.000.000s=sessionc=IN IP4 000.000.000.000t=0 0m=audio 18182 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -

4.4.4 GUEST →PBX

ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3afc8626;rport From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7aTo: <sip:[email protected]>;tag=as577af7ce Contact: <sip:[email protected]>Call-ID: [email protected]: 102 ACKUser-Agent: Asterisk PBX Max-Forwards: 70Content-Length: 0

41


4.4.5. GUEST ←PBX

BYE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;rport From: <sip:[email protected]>;tag=as577af7ceTo: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a Call-ID: [email protected]: 102 BYEUser-Agent: Asterisk PBX Max-Forwards: 70Content-Length: 0

4.4.6. GUEST →PBX

SIP/2.0 200 OKVia:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;received=000.000.000.000;rport=5060 From: <sip:[email protected]>;tag=as577af7ceTo: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a Call-ID: [email protected]: 102 BYEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesContact: <sip:[email protected]>Content-Length: 0

42


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,

figure 4.5 SIP flow including Busy message while incoming call


your IP-PBX000.000.000.000

IP address of SIP Trunk 2

server

1

2

3

4

CallerID

INVITEFrom: "080AAAAXXXX" <sip:080AAAAXXXX"@xxx.xxx.xxx.xxx>;tag=as0f1a5f0cTo: <sip:[email protected]>Call-ID: [email protected]

100 TryingFrom: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0cTo: <sip:[email protected]>Call-ID: [email protected]

486 Busy HereFrom: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0cTo: <sip:[email protected]>Call-ID: [email protected]

ACKFrom: " 080AAAAXXXX" " <sip:[email protected]>;tag=as0f1a5f0cTo: <sip:[email protected]>Call-ID: [email protected]

Recipient IP address of your IP-PBX

43


4.5.1 GUEST → PBX

INVITE sip:[email protected] SIP/2.0Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;rport From:" 080AAAAXXXX"<sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]>Contact: <sip: [email protected]>Call-ID: [email protected]: 102 INVITEUser-Agent: Asterisk PBXMax-Forwards: 70Date: Fri, 09 Jul 2010 02:27:46 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYSupported: replacesX-Asterisk-Guest-Tag: 00024X-Asterisk-Guest-Uniqueid: 1278642466.508Alert-info: 0312345678Content-Type: application/sdpContent-Length: 242

v=0o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxxs=sessionc=IN IP4 xxx.xxx.xxx.xxxt=0 0m=audio 10408 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv

4.5.2 PBX → GUEST

SIP/2.0 100 Trying Via: SIP/2.0/UDPxxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0cTo: <sip:[email protected]>Call-ID: [email protected]: 102 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:[email protected]> Content-Length: 0

44


4.5.3. PBX → GUEST

SIP/2.0 486 Busy Here Via: SIP/2.0/UDPxxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: " 080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0cTo: <sip:[email protected]>Call-ID: [email protected]: 102 INVITEContact: <sip:[email protected]> Content-Length: 0

4.5.4. GUEST→ PBX

Transmitting (NAT) to GUESTACK sip: [email protected] SIP/2.0Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;rport From:"080AAAAXXXX"<sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]>Contact: <sip:[email protected]>Call-ID: [email protected]: 102 ACKUser-Agent: Asterisk PBX Max-Forwards: 70Content-Length: 0

SIP Trunk 2 IP-PBX User Guide Asterisk - agileSIP Trunk 2 is a next generation IP phone service that...

Documents

Transcript of SIP Trunk 2 IP-PBX User Guide Asterisk - agileSIP Trunk 2 is a next generation IP phone service that...