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 USERNAME Password 1.2. On the left most column of the page, click SIP Trunk List.
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. Unique ID NAME Unique ID Name SIP TRUNK LIST LIST OF SIP TRUNK Channel (Number of Simultaneous call) Default: 2 Channels for Incoming & Outgoing NEXT: PURCHASE DID
1.5. From Circle Management Page, click Phone Number found at the leftmost column of the page. Phone Number List PHONE NUMBER: Phone list Buy / Purchase Phone Number (DID) Cancellation Phone Number Disturb Transmission Regulation Enter SIP and UID + User 1234567890joseSIP 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 THIS Click Search.
CLICK THIS 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
Go back to DID LIST (Phone LIST). DID NUMBER LIST Unique ID Associated with SIP (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)
Voipxxxx.xxxxx.xx To: <sip:0345900938@1.2.1.1> Incoming call s destination (CALLEE) number will also be displayed in Alert- info From: agile networks <sip:03450001280@113.34.235.106>;tag=as5dd4ea> 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. 200 201 Image 1. Organizational Chart of Incoming/Outgoing Calls
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) exten => _ XXX,, 3,Congestion exten => _ XXX,,104,Busy This rule is for dialing extension number. _XXX means 3 digit any number. 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) exten => _ XXX,, 3,Congestion exten => _ XXX,,104,Busy This rule is for dialing Extension number. _XXX means 3 digit any number. 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:password@voipxxxx.agile.ne.jp/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) exten => _ XXX,, 3,Congestion exten => _ XXX,,104,Busy This rule is for dialing Extension number. _XXX means 3 digit any number. 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: 0000111111@>;tag=as04bc6a95 To: <sip: 0000111111@> Call- ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1749 REGISTER Max- Forwards: 70 Expires: 120 Contact: <sip: 0000111111@1.2.1.1> Event: registration 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: 0000111111@>;tag=as04bc6a95 To: <sip: 0000111111@> Call- ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1749 REGISTER Supported: replaces Contact: <sip: 0000111111@> 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: 0000111111@>;tag=as04bc6a95 To: <sip: 0000111111@>;tag=as245298a3 Call- ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1749 REGISTER Supported: replaces WWW- Authenticate: Digest algorithm=md5, realm="voipxxxx.agile.ne.jp", nonce="3deff552"
3.1.4. PBX à GUEST REGISTER sip: SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;rport From: <sip: 0000111111@>;tag=as2031f6e2 To: <sip: 0000111111@> Call- ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1750 REGISTER 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: 0000111111@1.2.1.1> Event: registration 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 From: <sip: 0000111111@>;tag=as2031f6e2 To: <sip: 0000111111@> Call- ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1750 REGISTER Supported: replaces Contact: <sip: 0000111111@>
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 From: <sip: 0000111111@>;tag=as2031f6e2 To: <sip: 0000111111@>;tag=as245298a3 Call- ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1750 REGISTER Supported: replaces Expires: 120 Contact: <sip: 0000111111@1.2.1.1>;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:
CALLEE PBX USER 1.2.1.1 SET THE DISPLAY NAME FREELY CALLER ID GUEST SERVER 113.34.235.106 Guest Server IP Address START THE CONVERSATION TO END CALL Image 7. SIP message from PBX user to Agile s Guest Server during an outgoing call
3.2.1. PBX à GUEST INVITE sip:08058913782@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport From: "agile networks" <sip:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@> Contact: <sip:0345001280@1.2.1.1> Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 102 INVITE Max- Forwards: 70 Date: Fri, 02 Jul 2010 03:05:26 GMT 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: 0345001280@>;tag=as5dd4eaee To: <sip:08058913782@>;tag=as4abe0e65 Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 102 INVITE Supported: replaces
Proxy- Authenticate: Digest algorithm=md5, realm="voipxxxx.agile.ne.jp", nonce="23a44cfd" 3.2.3. PBX à GUEST ACK sip:08058913782@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport From: "agile networks" <sip:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@>;tag=as4abe0e65 Contact: <sip:0345001280@1.2.1.1> Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 102 ACK Max- Forwards: 70 3.2.4. PBX à GUEST INVITE sip:08058913782@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;rport From: "agile networks" <sip:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@> Contact: <sip:0345001280@1.2.1.1> Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 INVITE Max- Forwards: 70 Proxy- Authorization: Digest username="0000111111", realm="voipxxxx.agile.ne.jp", algorithm=md5, uri="sip:08058913782@", nonce="23a44cfd", response="cc6c5a668cbd435dee31c767981ff710", opaque="" Date: Fri, 02 Jul 2010 03:05:26 GMT 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:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@> Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@> 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:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@>;tag=as54380085 Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@>
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:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@>;tag=as54380085 Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@> Content- Type: application/sdp Content- Length: 242 v=0 o=root 4414 4414 IN IP4 s=session c=in IP4 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:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@>;tag=as54380085 Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@> Content- Type: application/sdp Content- Length: 242 v=0 o=root 4414 4415 IN IP4 s=session c=in IP4 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:08058913782@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK6c101c7f;rport From: "agile networks" <sip:0345001280@>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106>;tag=as54380085 Contact: <sip:0345001280@1.2.1.1> Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 ACK Max- Forwards: 70
3.2.10. GUEST à PBX BYE sip:0345001280@1.2.1.1 SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK166bf514;rport From: <sip:08058913782@>;tag=as54380085 To: "agile networks" <sip:0345001280@>;tag=as5dd4eaee Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 102 BYE Max- Forwards: 70 3.2.11. PBX à GUEST SIP/2.0 200 OK Via:SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK166bf514;received=;rport=5060 From: <sip:08058913782@>;tag=as54380085 To: "agile networks" <sip:0345001280@>;tag=as5dd4eaee Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 102 BYE Contact: <sip:0345001280@1.2.1.1> 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 CALLER ID CALLEE GUEST SERVER S IP ADDRESS Image 8. SIP Message when Callee is busy during an outgoing call from PBX User
3.3.1. PBX à GUEST INVITE sip:0345001028@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport From: "agile networks" <sip:0345001280@>;tag=as48ac6d56 To: <sip:0345001028@> Contact: <sip:0345001280@1.2.1.1> Call- ID: 1443bb69616709ff719769cc61d28ce0@ CSeq: 102 INVITE Max- Forwards: 70 Date: Tue, 06 Jul 2010 10:09:37 GMT 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:0345001028@>;tag=as291aca90 Call- ID: 1443bb69616709ff719769cc61d28ce0@ CSeq: 102 INVITE Supported: replaces Proxy- Authenticate: Digest algorithm=md5, realm="voipxxxx.agile.ne.jp", nonce="15a6e863"
3.3.3. PBX à Guest ACK sip:0345001028@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport From: "agile networks" <sip:0345001280@>;tag=as48ac6d56 To: <sip:0345001028@>;tag=as291aca90 Contact: <sip:0345001280@1.2.1.1> Call- ID: 1443bb69616709ff719769cc61d28ce0@ CSeq: 102 ACK Max- Forwards: 70 3.3.4. PBX à GUEST INVITE sip:0345001028@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" <sip:0345001280@>;tag=as48ac6d56 To: <sip:0345001028@> Contact: <sip:0345001280@1.2.1.1> Call- ID: 1443bb69616709ff719769cc61d28ce0@ CSeq: 103 INVITE Max- Forwards: 70 Proxy- Authorization: Digest username="0000185475", realm="voipxxxx.agile.ne.jp", algorithm=md5, uri="sip:0345001028@", nonce="15a6e863", response="54ebd3bdb5bab4b621f55fbd3ffe5e0b", opaque="" Date: Tue, 06 Jul 2010 10:09:37 GMT 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:0345001280@>;tag=as48ac6d56 To: <sip:0345001028@> Call- ID: 1443bb69616709ff719769cc61d28ce0@ CSeq: 103 INVITE low: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:0345001028@> 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:0345001280@>;tag=as48ac6d56 To: <sip:0345001028@>;tag=as715c3c5e Call- ID: 1443bb69616709ff719769cc61d28ce0@ CSeq: 103 INVITE Contact: <sip:0345001028@> 3.3.7. PBX à GUEST ACK sip:0345001028@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" <sip:0345001280@>;tag=as48ac6d56 To: <sip:0345001028@>;tag=as715c3c5e Contact: <sip:0345001280@1.2.1.1> Call- ID: 1443bb69616709ff719769cc61d28ce0@ CSeq: 103 ACK Max- Forwards: 70
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: PBX USER 1.2.1.1 Caller ID Guest Server Destination Guest Server IP Address IP Address PBX Start the Conversation To end call Image 9: SIP Message from Guest Server to PBX user during an Incoming Call
3.4.1. GUEST à PBX INVITE sip:0345900938@1.2.1.1 SIP/2.0 Via: SIP/2.0/UDP 1 :5060;branch=z9hG4bK546a1def;rport From: "08058913782" <sip:08058913782@>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1> Contact: <sip:08058913782@> Call- ID: 490e49cf2141339f0007e5ce47d80dd1@ CSeq: 102 INVITE Max- Forwards: 70 Date: Fri, 02 Jul 2010 05:41:33 GMT 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 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=;rport=5060 From: "08058913782" <sip:08058913782@>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1> Call- ID: 490e49cf2141339f0007e5ce47d80dd1@ CSeq: 102 INVITE
Contact: <sip:0345900938@1.2.1.1> 3.4.3. GUEST ß PBX SIP/2.0 200 OK Via:SIP/2.0/UDP 13.34.235.106:5060;branch=z9hG4bK546a1def;received=;rport=5060 From: "08058913782" <sip:08058913782@>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1>;tag=as577af7ce Call- ID: 490e49cf2141339f0007e5ce47d80dd1@ CSeq: 102 INVITE Contact: <sip:0345900938@1.2.1.1> 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:0345900938@1.2.1.1 SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK3afc8626;rport From: "08058913782" <sip:08058913782@>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1>;tag=as577af7ce Contact: <sip:08058913782@> Call- ID: 490e49cf2141339f0007e5ce47d80dd1@ CSeq: 102 ACK Max- Forwards: 70
3.4.5. GUEST ß PBX BYE sip:08058913782@ SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK5b3130a7;rport From: <sip:0345900938@1.2.1.1>;tag=as577af7ce To: "08058913782" <sip:08058913782@>;tag=as1dddca7a Call- ID: 490e49cf2141339f0007e5ce47d80dd1@ CSeq: 102 BYE Max- Forwards: 70 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:0345900938@1.2.1.1>;tag=as577af7ce To: "08058913782" <sip:08058913782@>;tag=as1dddca7a Call- ID: 490e49cf2141339f0007e5ce47d80dd1@ CSeq: 102 BYE Supported: replaces Contact: <sip:08058913782@>
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: PBX USER 1.2.1.1 CALLER ID GUEST SERVER CALLEE GUEST SERVER S IP ADDRESS PBX S IP ADDRESS Image 10. SIP message from Guest Server to PBX user during an Incoming Call 3.5.1. GUEST à PBX INVITE sip:0345900938@1.2.1.1 SIP/2.0 Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;rport From: "0345900846" <sip:0345900846@>;tag=as0f1a5f0c To: <sip:0345900938@1.2.1.1> Contact: <sip:0345900846@> Call- ID: 1aa4d60711e0817d731834f474d958b0@ CSeq: 102 INVITE
Max- Forwards: 70 Date: Fri, 09 Jul 2010 02:27:46 GMT 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 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=;rport=5060 From: "0345900846" <sip:0345900846@>;tag=as0f1a5f0c To: <sip:0345900938@1.2.1.1> Call- ID: 1aa4d60711e0817d731834f474d958b0@ CSeq: 102 INVITE Contact: <sip:0345900938@1.2.1.1> 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=;rport=5060 From: "0345900846" <sip:0345900846@>;tag=as0f1a5f0c To: <sip:0345900938@1.2.1.1> Call- ID: 1aa4d60711e0817d731834f474d958b0@ CSeq: 102 INVITE Contact: <sip:0345900938@1.2.1.1> 3.5.4. GUEST à PBX Transmitting (NAT) to GUEST ACK sip: 0345900938@ SIP/2.0 Via: SIP/2.0/UDP :5060;branch= z9hg4bk0b7fb7b8;rport From: "0345900846" <sip:0345900846@>;tag=as0f1a5f0c To: <sip:0345900938@1.2.1.1> Contact: <sip:0345900846@1.2.1.1> Call- ID: 6dd7b12f1438e1572cae057f274419e6@1.2.1.1 CSeq: 102 ACK Max- Forwards: 70