AGILESIPTRUNK! IP.PBXConnectionManual! (Asterisk,Trixbox)! · [200]! type=friend! username=645! secret=645pass! host=dynamic! context=outbound.1!! [201]! type=friend! username=646!

AGILE SIP TRUNK IP-‐PBX Connection Manual (Asterisk, Trixbox)

1. SIP TRUNK SETTINGS 1.1. Login to CID (Customer ID): https://manager.agile.ne.jp/login.php

1.2. On the left most column of the page, click SIP Trunk List.

USERNAME

Password

1.3. On the upper portion of the page, move the mouse over the Purchase/Terminate tab and click Purchase SIP Trunk.

On Purchase SIP Trunk page, select one item for each: SIP Trunk and Additional Channel SIP Trunk. Then, click Add to Cart. Click Next. Modify your purchase by checking and unchecking the row/s of items to purchase. Click Next. Then, click Purchase.

1.4. Go to SIP TRUNK LIST.

NEXT: PURCHASE DID

SIP TRUNK LIST LIST OF SIP TRUNK

Unique ID Name Unique ID NAME

Channel (Number of Simultaneous call) Default: 2 Channels for Incoming & Outgoing

1.5. From Circle Management Page, click Phone Number found at the leftmost column of the page.

PHONE NUMBER:

• Phone list • Buy / Purchase Phone Number (DID) • Cancellation Phone Number • Disturb • Transmission Regulation

Move the mouse over the Purchase/Terminate tab found at the upper part of the page to display selections. On the selections, click Purchase Phone Number.

Click Search.

Phone Number List

CLICK THIS

Enter SIP and UID + User 1234567890joseSIP

BUY PHONE NUMBER Choose Provider (KDDI, NTT) or search using Area Code. Tick the check box opposite the preferred phone number. Click Add to Cart.

AREA CODE

CLICK THIS

Go back to DID LIST (Phone LIST).

(Purchased DID is listed here.) *Configuring Agile Phone for SIP Trunk is possible. Note: Unique ID can be used with multiple DID. Ex: UID DID 1234567890 => 0345131495; 0368302379; 0671763839 1.7 . AGILE SIP TRUNK

Agile SIP Trunk, service that assigns multiple phone numbers (DID) and number of multiple call (channels) with only one Unique ID (SIP user account). By using SIP Trunk, it is possible to easily execute external line connection to a main device that supports SIP and representative PBX software. ATTENTION � One assigned Unique ID for one PBX user. � Support for operability validated previous versions is not executed. Operability Validated: IP-‐PBX Asterisk version: 1.6.2.9 Trixbox version: PBXtra core fon_p_1.2.17_JP EXAMPLE OF CONFIGURATION Unique that is registered in Agile’s Guest Server: 0000185475 Login Server (Agile’s Guest Server): voip3024.agile.ne.jp (113.34.235.106) PBX User: 1.2.1.1 Outgoing call’s originator (CALLER): 0349000938, 03450001280 Outgoing call’s originator (CALLER): agile networks (can be set freely) Incoming call’s destination (CALLER): 0345900938, 0345001280 SIP Extension Line; 2 devices (200-‐201)

DID NUMBER LIST Unique ID Associated with SIP

To: <sip:[email protected]> From: “agile networks” Incoming call’s destination (CALLEE) number will also be displayed in Alert-‐info <sip:[email protected]>;tag=as5dd4ea>

200 201

Image 1. Organizational Chart of Incoming/Outgoing Calls

Refer to 4.1 of table of contents for details set in SIP message’s To Header in incoming call DID during an incoming call.

Refer to 2.1 of table of contents for details set in SIP message’s From Header in Outgoing caller’s number during an incoming call.

Voipxxxx.xxxxx.xx

2. SETTING EXAMPLE 2.1. SETTING OF A SAMPLE ACCOUNT IN ASTERISK: Unique ID: UID Password: “Your password” Incoming call’s destination(CALLEE): DID1, DID2 Outgoing caller’s number: DID1, DID2 Login Server: voip3024.agile.ne.jp Example of SIP extension (645-‐646) and Agile SIP trunk

• Incoming call’s destination (CALLEE) DID: the case of "DID1", call will be placed to extension number "645"

• Incoming call’s destination (CALLEE) DID: the case of "DID2", call will be placed to extension number "646"

• During an outgoing call from "645", outgoing caller number (CALLER ID) is set to "DID1" and the outgoing call is placed.

• During an outgoing call from "646", outgoing caller number (CALLER ID) is set to "DID2” and the outgoing call is placed.

-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ sip.conf -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp register => UID:password@siptr [siptr] type=friend username=UID secret=password context=inbound canreinvite=no host=voipxxxx.agile.ne.jp insecure=port,invite disallow=all allow=ulaw Continue………

[200] type=friend username=645 secret=645pass host=dynamic context=outbound-‐1 [201] type=friend username=646 secret=646pass host=dynamic context=outbound-‐2 -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ extensions.conf -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] writeprotect=no priorityjumping=yes [inbound] ;exten => Incoming Call’s Destination (CALLEE) DID, 1,Dial(SIP/CALLEE’S EXTENSION NUMBER,120,t) ;exten => Incoming Call’s Destination (CALLEE) DID, 2,Congestion ;exten => Incoming Call’s Destination (CALLEE) DID,102,Busy exten => DID1, 1,Dial(SIP/645,120,t) exten => DID1, 2,Congestion exten => DID1,102,Busy exten => DID2, 1,Dial(SIP/646,120,t) exten => DID2, 2,Congestion exten => DID2,102,Busy ;[outbound] ;exten => _0., 1,Set(CALLERID(num)=Caller ID) ;exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) ;exten => _0., 3,Congestion ;exten => _0.,103,Busy [outbound-‐1] exten => _ XXX, 1,Set(CALLERID(num)= DID1) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301

exten => _0., 1,Set(CALLERID(num)= DID1) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Congestion exten => _0.,104,Busy [outbound-‐2] exten => _ XXX, 1,Set(CALLERID(num)= DID2) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing Extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301 exten => _0., 1,Set(CALLERID(num)= DID2) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Congestion exten => _0.,104,Busy 2.2. Configuration example to limit the number of simultaneous calls for each group in Asterisk

• Group 1: Numbers of Simultaneous Calls “Limit: 2”; Extensions 201~202; Phone Number: 0345131495 • Group 2: Numbers of Simultaneous Calls “Limit: 3”; Extensions 301~302; Phone Number: 0344368713 • Unique ID registered to Agile’s Guest Server: UID • Login server (Agile’s Guest Server): voipXXXX.agile.ne.jp

-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ sip.conf -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [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=>UID:[email protected]/SIPTR [SIPTR] type=friend username=0000222222 secret=password host= voipqwer.agile.ne.jp

context=inbound ; Extensions of 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 ; Extensions of 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 -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ extensions.conf -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] writeprotect=no priorityjumping=yes ; Example of Channel Limit (Incoming Call) [inbound] ; Group 1 exten => 0333333333, 1,NoOp(EXTEN: ${EXTEN}) exten => 0333333333, 2,Set(GROUP(CALLS)=GROUP1) exten => 0333333333, 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0333333333, 4,Set(MAXCALLS=2) exten => 0333333333, 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] | Hangup) exten => 0333333333, 6,Dial(SIP/201&SIP/202,120) exten => 0333333333, 7,Congestion exten => 0333333333,106,Busy

; Group 2 exten => 0333333333, 1,NoOp(EXTEN: ${EXTEN}) exten => 0333333333, 2,Set(GROUP(CALLS)=GROUP1) exten => 0333333333, 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0333333333, 4,Set(MAXCALLS=3) exten => 0333333333, 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] | Hangup) exten => 0333333333, 6,Dial(SIP/301&SIP/302,120) exten => 0333333333, 7,Congestion exten => 0333333333,106,Busy ; Example of Channel Limit (Outgoing Call) ; Group 1 [group1_outbound] exten => _ XXX, 1,Set(CALLERID(num)= 0345131495) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing Extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301 exten => _0., 1,Set(CALLERID(num)= 0345131495) 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}@SIPTR,120) exten => _0., 8,Congestion exten => _0.,106,Busy ; Group 2 [group2_outbound] exten => _XXX, 1,Set(CALLERID(num)= 0344368713) exten => _XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _ XXX, 3,Congestion exten => _ XXX ,104,Busy exten => _0., 1,Set(CALLERID(num)= 0344368713) exten => _0., 2,Set(CALLERID(name)=GROUP2) exten => _0., 3,Set(GROUP(CALLS)=GROUP2) exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => _0., 5,Set(MAXCALLS=3) exten => _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] | Hangup) exten => _0., 7,Dial(SIP/${EXTEN}@SIPTR,120) exten => _0., 8,Congestion exten => _0.,106,Busy

ATTENTION: “|” will become “?” for Asterisk ver1.42 or lower.

2.3 SETTINGS IN TRIXBOX

2.3. Example of Account Setting in Trixbox

2.3.1. Example of Unique ID Setting

Image 2. Example of Unique ID Setting

2.3.2. Example of Phone Number/User PBX Extension Line Setting

Image 3. Example of PBX Extension Number/User Setting

2.3.3. Phone Number/User PBX Setting Extension Line Setting Example

During an incoming call to a Callee’s DID 03450001280, extension line 5001 will be called When making an outgoing call from extension line 5001, set 0350001280 in Outgoing Call Number and a call is placed.

Image 4. User PBX Extension Line (5001)’s Setting

During an incoming call to a Callee’s DID 03450001280, extension line 5002 will be called When making an outgoing call from extension line 5002, set 0350001280 in CALLER ID and a call is placed.

Image 5. PBX User: Extension Number (5002)’s Setting

3. Technical Data

3.1. SIP message when registering the user's information to the guest PBX server:

Authenticates the user's PBX to the guest server and registers the address information and the Unique ID information.

Examples of SIP messages are as follows:

PBX USER 1.2.1.1 GUEST SERVER

113.34.235.106

UNIQUE ID TO REGISTER IN AGILE’S GUEST SERVER GUEST SERVER’S IP

ADDRESS

Image 6. SIP Message during registration of PBX user’s information to Guest Server

3.1.1. PBX à GUEST REGISTER sip:113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4e9b3e05;rport From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-‐ID: [email protected] CSeq: 1749 REGISTER User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Expires: 120 Contact: <sip: [email protected]> Event: registration Content-‐Length: 0 3.1.2. GUEST à PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4e9b3e05;received=1.2.1.1;rport=5060 From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-‐ID: [email protected] CSeq: 1749 REGISTER User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip: [email protected]> Content-‐Length: 0 3.1.3. GUESTà PBX SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4e9b3e05;received=1.2.1.1;rport=5060 From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]>;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="voipxxxx.agile.ne.jp", nonce="3deff552" Content-‐Length: 0

3.1.4. PBX à GUEST REGISTER sip: 000.34.235.106 SIP/2.0

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;rport

From: <sip: [email protected]>;tag=as2031f6e2

To: <sip: [email protected]>

Call-‐ID: [email protected]

CSeq: 1750 REGISTER

User-‐Agent: Asterisk PBX

Max-‐Forwards: 70

Authorization: Digest username="0000111111", realm="voipxxxx.agile.ne.jp", algorithm=MD5,

uri="sip: 113.34.235.106", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056",

opaque=""

Expires: 120

Contact: <sip: [email protected]>

Event: registration

Content-‐Length: 0

3.1.5. GUEST à PBX SIP/2.0 100 Trying

Via:SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;received=1.2.1.1;rport=5060


To: <sip: [email protected]>


CSeq: 1750 REGISTER


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

Supported: replaces

Contact: <sip: [email protected]>


3.1.6. GUEST à PBX

SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;received=1.2.1.1;rport=5060


To: <sip: [email protected]>;tag=as245298a3


CSeq: 1750 REGISTER


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

Supported: replaces

Expires: 120

Contact: <sip: [email protected]>;expires=120

Date: Mon, 05 Jul 2010 04:20:13 GMT


3.2. During an outgoing calling from PBX User to Guest Server:

§ On PBX User, set the outgoing caller number (Caller ID) in From Header.

§ Field value for From Header’s name can be set freely.

From: "name" <sip: Caller ID@Guest Server IP address or Domain Name＞

§ Examples of SIP messages are as follows:

PBX USER 1.2.1.1

GUEST SERVER 113.34.235.106

Guest Server IP Address

SET THE DISPLAY NAME FREELY CALLER ID

CALLEE

START THE CONVERSATION

TO END CALL

Image 7. SIP message from PBX user to Agile’s Guest Server during an outgoing call

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

3.2.1. PBX à GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] 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 1.2.1.1 s=session c=IN IP4 1.2.1.1 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 -‐ -‐ -‐ -‐ 3.2.2. GUESTà PBX SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;received=1.2.1.1;rport=5060 From: " agile networks " <sip: [email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces

Proxy-‐Authenticate: Digest algorithm=MD5, realm="voipxxxx.agile.ne.jp", nonce="23a44cfd" Content-‐Length: 0 3.2.3. PBX à GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 3.2.4. PBX à GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;rport From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Proxy-‐Authorization: Digest username="0000111111", realm="voipxxxx.agile.ne.jp", algorithm=MD5, uri="sip:[email protected]", 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 1.2.1.1 s=session c=IN IP4 1.2.1.1 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 -‐ -‐ -‐ -‐ 3.2.5. GUEST à PBX SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;received=1.2.1.1;rport=5060 From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0 3.2.6. GUEST à PBX SIP/2.0 180 Ringing Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;received=1.2.1.1;rport=5060 From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0

3.2.7. GUEST à PBX SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;received=1.2.1.1;rport=5060 From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Type: application/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 000.34.235.106 s=session c=IN IP4 000.34.235.106 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

3.2.8. GUEST à PBX SIP/2.0 200 OK Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;received=1.2.1.1;rport=5060 From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Type: application/sdp Content-‐Length: 242 v=0 o=root 4414 4415 IN IP4 000.34.235.106 s=session c=IN IP4 000.34.235.106 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 3.2.9. PBX à GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK6c101c7f;rport From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0

3.2.10. GUEST à PBX BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.34.235.106:5060;branch=z9hG4bK166bf514;rport From: <sip:[email protected]>;tag=as54380085 To: "agile networks" <sip:[email protected]>;tag=as5dd4eaee Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 3.2.11. PBX à GUEST SIP/2.0 200 OK Via:SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK166bf514;received=000.34.235.106;rport=5060 From: <sip:[email protected]>;tag=as54380085 To: "agile networks" <sip:[email protected]>;tag=as5dd4eaee Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Length: 0 X-‐Asterisk-‐HangupCause: Normal Clearing

3.3. PBX User in case the incoming call destination (CALLEE) was busy when making calls SIP message:

� After an outgoing call from PBX user, if the incoming call destination (CALLEE) is still unreachable, Busy Here message is sent from Guest server to the PBX user.

� During an incoming call from PBX user, examples of SIP messages if the incoming call destination (CALLEE) is still busy are as follows:

PBX USER 1.2.1.1 GUEST SERVER

000.34.235.106

GUEST SERVER’S IP ADDRESS

CALLER ID

CALLEE

Image 8. SIP Message when Callee is busy during an outgoing call from PBX User

000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106 000.34.235.106

000.34.235.106

3.3.1. PBX à GUEST

INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport From: "agile networks" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] 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 1.2.1.1 s=session c=IN IP4 1.2.1.1 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 -‐ -‐ -‐ -‐

3.3.2. GUESTà PBX

SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;received=1.2.1.1;rport=5060 To: <sip:[email protected]>;tag=as291aca90 Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-‐Authenticate: Digest algorithm=MD5, realm="voipxxxx.agile.ne.jp", nonce="15a6e863" Content-‐Length: 0

3.3.3. PBX à Guest ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport From: "agile networks" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as291aca90 Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 3.3.4. PBX àGUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Proxy-‐Authorization: Digest username="0000185475", realm="voipxxxx.agile.ne.jp", algorithm=MD5, uri="sip:[email protected]", 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 1.2.1.1 s=session c=IN IP4 1.2.1.1 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 -‐ -‐ -‐ -‐ 3.3.5. GUEST à PBX SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;received=1.2.1.1;rport=5060 From: "agile networks" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX low: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0 3.3.6. GUEST à PBX SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;received=1.2.1.1;rport=5060 From: "agile networks" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Call-‐ID: [email protected] CSeq: 103 INVITE User-‐Agent: Asterisk PBX Contact: <sip:[email protected]> Content-‐Length: 0 3.3.7. PBX à GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 103 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0

3.4. When coming from the guest PBX server to the user:

§ Set incoming call destination (CALLEE) in To Header and Alert Info Header for the Guest Server. To: <sip: Destination (CALLEE) phone number@PBX user IP Address>

§ Examples of SIP messages are as follows:

Image 9: SIP Message from Guest Server to PBX user during an Incoming Call

PBX USER 1.2.1.1

Caller ID

Guest Server 000.34.235.106

Destination Guest Server IP Address

IP Address PBX

Start the Conversation

To end call

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

3.4.1. GUEST à PBX INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1 000.34.235.106:5060;branch=z9hG4bK546a1def;rport From: "08058913782" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]> 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: 0345900938 Content-‐Type: application/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 000.34.235.106 s=session c=IN IP4 113.34.235.106 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 3.4.2. GUEST ß PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK546a1def;received=000.34.235.106;rport=5060 From: "08058913782" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE

User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Length: 0 3.4.3. GUEST ß PBX SIP/2.0 200 OK Via:SIP/2.0/UDP 13.34.235.106:5060;branch=z9hG4bK546a1def;received=000.34.235.106;rport=5060 From: "08058913782" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Type: application/sdp Content-‐Length: 220 v=0 o=root 22702 22702 IN IP4 1.2.1.1 s=session c=IN IP4 1.2.1.1 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 -‐ -‐ -‐ -‐ 3.4.4. GUEST à PBX ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.34.235.106:5060;branch=z9hG4bK3afc8626;rport From: "08058913782" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70

Content-‐Length: 0 3.4.5. GUEST ß PBX BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK5b3130a7;rport From: <sip:[email protected]>;tag=as577af7ce To: "08058913782" <sip:[email protected]>;tag=as1dddca7a Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 3.4.6. GUEST à PBX SIP/2.0 200 OK Via:SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK5b3130a7;received=1.2.1.1;rport=5060 From: <sip:[email protected]>;tag=as577af7ce To: "08058913782" <sip:[email protected]>;tag=as1dddca7a Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-‐Length: 0

3.5. From Guest Server to PBX user during an incoming call

§ Set incoming call destination (CALLEE) in To Header and Alert Info Header for the Guest Server.

To: <sip: Destination (CALLEE) phone number@PBX user IP Address> § Examples of SIP messages are as follows:

Image 10. SIP message from Guest Server to PBX user during an Incoming Call

3.5.1. GUEST à PBX INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;rport From: "0345900846" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE

PBX USER 1.2.1.1 CALLER ID

GUEST SERVER 000.34.235.106

GUEST SERVER’S IP ADDRESS

PBX’S IP ADDRESS

CALLEE 000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

000.34.235.106

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: 0345900938 Content-‐Type: application/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 000.34.235.106 s=session c=IN IP4 113.34.235.106 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 3.5.2. PBX à GUEST SIP/2.0 100 Trying Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;received=000.34.235.106;rport=5060 From: "0345900846" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE User-‐Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[email protected]> Content-‐Length: 0 3.5.3. PBX à GUEST SIP/2.0 486 Busy Here

Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;received=000.34.235.106;rport=5060 From: "0345900846" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE Contact: <sip:[email protected]> Content-‐Length: 0 3.5.4. GUEST à PBX Transmitting (NAT) to GUEST ACK sip: [email protected] SIP/2.0 Via: SIP/2.0/UDP 000.34.235.106:5060;branch= z9hG4bK0b7fb7b8;rport From: "0345900846" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Contact: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 ACK User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0

Documents

AGILESIPTRUNK! IP.PBXConnectionManual! (Asterisk,Trixbox)! · [200]! type=friend! username=645! secret=645pass! host=dynamic! context=outbound.1!! [201]! type=friend! username=646!