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

Transcription

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

2 1. SIP TRUNK SETTINGS 1.1. Login to CID (Customer ID): USERNAME Password 1.2. On the left most column of the page, click SIP Trunk List.

3 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.

4 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

5 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 joseSIP 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.

6 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

7 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 => ; ; 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: Trixbox version: PBXtra core fon_p_1.2.17_jp EXAMPLE OF CONFIGURATION Unique that is registered in Agile s Guest Server: Login Server (Agile s Guest Server): voip3024.agile.ne.jp ( ) PBX User: Outgoing call s originator (CALLER): , Outgoing call s originator (CALLER): agile networks (can be set freely) Incoming call s destination (CALLER): , SIP Extension Line; 2 devices ( )

8 Voipxxxx.xxxxx.xx To: Incoming call s destination (CALLEE) number will also be displayed in Alert- info From: agile networks <sip: @ >;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 Image 1. Organizational Chart of Incoming/Outgoing Calls

9 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 ( ) 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= 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

10 [200] type=friend username=645 secret=645pass host=dynamic context=outbound- 1 [201] type=friend username=646 secret=646pass host=dynamic context=outbound 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

11 exten => _0., 1,Set(CALLERID(num)= DID1) exten => _0., 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: Group 2: Numbers of Simultaneous Calls Limit: 3 ; Extensions 301~302; Phone Number: 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= srvlookup=yes disallow=all allow=ulaw language=jp register=>uid:password@voipxxxx.agile.ne.jp/siptr [SIPTR] type=friend username= secret=password host= voipqwer.agile.ne.jp

12 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 => , 1,NoOp(EXTEN: ${EXTEN}) exten => , 2,Set(GROUP(CALLS)=GROUP1) exten => , 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => , 4,Set(MAXCALLS=2) exten => , 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] Hangup) exten => , 6,Dial(SIP/201&SIP/202,120) exten => , 7,Congestion exten => ,106,Busy

13 ; Group 2 exten => , 1,NoOp(EXTEN: ${EXTEN}) exten => , 2,Set(GROUP(CALLS)=GROUP1) exten => , 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => , 4,Set(MAXCALLS=3) exten => , 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] Hangup) exten => , 6,Dial(SIP/301&SIP/302,120) exten => , 7,Congestion exten => ,106,Busy ; Example of Channel Limit (Outgoing Call) ; Group 1 [group1_outbound] exten => _ XXX, 1,Set(CALLERID(num)= ) exten => _ XXX, 2,Dial(SIP/${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)= ) 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)= ) exten => _XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _ XXX, 3,Congestion exten => _ XXX,104,Busy exten => _0., 1,Set(CALLERID(num)= ) exten => _0., 2,Set(CALLERID(name)=GROUP2) exten => _0., 3,Set(GROUP(CALLS)=GROUP2) exten => _0., 4,Set(CURRENTCALLS=${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.

14 2.3 SETTINGS IN TRIXBOX 2.3. Example of Account Setting in Trixbox Example of Unique ID Setting Image 2. Example of Unique ID Setting

15 Example of Phone Number/User PBX Extension Line Setting Image 3. Example of PBX Extension Number/User Setting

16 Phone Number/User PBX Setting Extension Line Setting Example During an incoming call to a Callee s DID , extension line 5001 will be called When making an outgoing call from extension line 5001, set in Outgoing Call Number and a call is placed. Image 4. User PBX Extension Line (5001) s Setting

17 During an incoming call to a Callee s DID , extension line 5002 will be called When making an outgoing call from extension line 5002, set in CALLER ID and a call is placed. Image 5. PBX User: Extension Number (5002) s Setting

18 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 GUEST SERVER 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

19 PBX à GUEST REGISTER sip: SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK4e9b3e05;rport From: <sip: To: <sip: Call- ID: CSeq: 1749 REGISTER Max- Forwards: 70 Expires: 120 Contact: <sip: Event: registration GUEST à PBX SIP/ Trying Via:SIP/2.0/UDP :5060;branch=z9hG4bK4e9b3e05;received= ;rport=5060 From: <sip: To: <sip: Call- ID: CSeq: 1749 REGISTER Supported: replaces Contact: <sip: GUESTà PBX SIP/ Unauthorized Via: SIP/2.0/UDP :5060;branch=z9hG4bK4e9b3e05;received= ;rport=5060 From: <sip: To: <sip: Call- ID: CSeq: 1749 REGISTER Supported: replaces WWW- Authenticate: Digest algorithm=md5, realm="voipxxxx.agile.ne.jp", nonce="3deff552"

20 PBX à GUEST REGISTER sip: SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1db71efa;rport From: <sip: To: <sip: Call- ID: CSeq: 1750 REGISTER Max- Forwards: 70 Authorization: Digest username=" ", realm="voipxxxx.agile.ne.jp", algorithm=md5, uri="sip: ", nonce="3deff552", response="bace343abbe dba84e58d7e056", opaque="" Expires: 120 Contact: <sip: Event: registration GUEST à PBX SIP/ Trying Via:SIP/2.0/UDP :5060;branch=z9hG4bK1db71efa;received= ;rport=5060 From: <sip: To: <sip: Call- ID: CSeq: 1750 REGISTER Supported: replaces Contact: <sip:

21 GUEST à PBX SIP/ OK Via: SIP/2.0/UDP :5060;branch=z9hG4bK1db71efa;received= ;rport=5060 From: <sip: To: <sip: Call- ID: CSeq: 1750 REGISTER Supported: replaces Expires: 120 Contact: <sip: Date: Mon, 05 Jul :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:

22 CALLEE PBX USER SET THE DISPLAY NAME FREELY CALLER ID GUEST SERVER 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

23 PBX à GUEST INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK17bf4505;rport From: "agile networks" To: Contact: Call- ID: CSeq: 102 INVITE Max- Forwards: 70 Date: Fri, 02 Jul :05:26 GMT Content- Type: application/sdp Content- Length: 267 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off GUESTà PBX SIP/ Proxy Authentication Required Via: SIP/2.0/UDP :5060;branch=z9hG4bK17bf4505;received= ;rport=5060 From: " agile networks " <sip: @>;tag=as5dd4eaee To: <sip: @>;tag=as4abe0e65 Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 102 INVITE Supported: replaces

24 Proxy- Authenticate: Digest algorithm=md5, realm="voipxxxx.agile.ne.jp", nonce="23a44cfd" PBX à GUEST ACK SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK17bf4505;rport From: "agile networks" To: Contact: Call- ID: CSeq: 102 ACK Max- Forwards: PBX à GUEST INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;rport From: "agile networks" To: Contact: Call- ID: CSeq: 103 INVITE Max- Forwards: 70 Proxy- Authorization: Digest username=" ", realm="voipxxxx.agile.ne.jp", algorithm=md5, nonce="23a44cfd", response="cc6c5a668cbd435dee31c767981ff710", opaque="" Date: Fri, 02 Jul :05:26 GMT Content- Type: application/sdp Content- Length: 267 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000

25 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off GUEST à PBX SIP/ Trying Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "agile networks" <sip: @>;tag=as5dd4eaee To: <sip: @> Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 INVITE Supported: replaces Contact: <sip: @> GUEST à PBX SIP/ Ringing Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "agile networks" <sip: @>;tag=as5dd4eaee To: <sip: @>;tag=as Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 INVITE Supported: replaces Contact: <sip: @>

26 GUEST à PBX SIP/ Session Progress Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "agile networks" To: Call- ID: CSeq: 103 INVITE Supported: replaces Contact: Content- Type: application/sdp Content- Length: 242 v=0 o=root IN IP4 s=session c=in IP4 t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=ptime:20 a=sendrecv

27 GUEST à PBX SIP/ OK Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "agile networks" To: Call- ID: CSeq: 103 INVITE Supported: replaces Contact: Content- Type: application/sdp Content- Length: 242 v=0 o=root IN IP4 s=session c=in IP4 t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=ptime:20 a=sendrecv PBX à GUEST ACK sip: @ SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK6c101c7f;rport From: "agile networks" <sip: @>;tag=as5dd4eaee To: <sip: @ >;tag=as Contact: <sip: @ > Call- ID: 6426c31c421e503b72515b46569f2ee0@ CSeq: 103 ACK Max- Forwards: 70

28 GUEST à PBX BYE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK166bf514;rport From: To: "agile networks" Call- ID: CSeq: 102 BYE Max- Forwards: PBX à GUEST SIP/ OK Via:SIP/2.0/UDP :5060;branch=z9hG4bK166bf514;received=;rport=5060 From: To: "agile networks" Call- ID: CSeq: 102 BYE Contact: X- Asterisk- HangupCause: Normal Clearing

29 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 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

30 PBX à GUEST INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK63c44c39;rport From: "agile networks" To: Contact: Call- ID: CSeq: 102 INVITE Max- Forwards: 70 Date: Tue, 06 Jul :09:37 GMT Content- Type: application/sdp Content- Length: 267 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off GUESTà PBX SIP/ Proxy Authentication Required Via: SIP/2.0/UDP :5060;branch=z9hG4bK63c44c39;received= ;rport=5060 To: <sip: @>;tag=as291aca90 Call- ID: 1443bb ff719769cc61d28ce0@ CSeq: 102 INVITE Supported: replaces Proxy- Authenticate: Digest algorithm=md5, realm="voipxxxx.agile.ne.jp", nonce="15a6e863"

31 PBX à Guest ACK SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK63c44c39;rport From: "agile networks" To: Contact: Call- ID: CSeq: 102 ACK Max- Forwards: PBX à GUEST INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" To: Contact: Call- ID: CSeq: 103 INVITE Max- Forwards: 70 Proxy- Authorization: Digest username=" ", realm="voipxxxx.agile.ne.jp", algorithm=md5, nonce="15a6e863", response="54ebd3bdb5bab4b621f55fbd3ffe5e0b", opaque="" Date: Tue, 06 Jul :09:37 GMT Content- Type: application/sdp Content- Length: 267 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000

32 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off GUEST à PBX SIP/ Trying Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;received= ;rport=5060 From: "agile networks" <sip: @>;tag=as48ac6d56 To: <sip: @> Call- ID: 1443bb ff719769cc61d28ce0@ CSeq: 103 INVITE low: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip: @> GUEST à PBX SIP/ Busy Here Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;received= ;rport=5060 From: "agile networks" <sip: @>;tag=as48ac6d56 To: <sip: @>;tag=as715c3c5e Call- ID: 1443bb ff719769cc61d28ce0@ CSeq: 103 INVITE Contact: <sip: @> PBX à GUEST ACK sip: @ SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" <sip: @>;tag=as48ac6d56 To: <sip: @>;tag=as715c3c5e Contact: <sip: @ > Call- ID: 1443bb ff719769cc61d28ce0@ CSeq: 103 ACK Max- Forwards: 70

33 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 user IP Address> Examples of SIP messages are as follows: PBX USER 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

34 GUEST à PBX INVITE SIP/2.0 Via: SIP/2.0/UDP 1 :5060;branch=z9hG4bK546a1def;rport From: " " <sip: @>;tag=as1dddca7a To: <sip: @ > Contact: <sip: @> Call- ID: 490e49cf f0007e5ce47d80dd1@ CSeq: 102 INVITE Max- Forwards: 70 Date: Fri, 02 Jul :41:33 GMT Supported: replaces X- Asterisk- Guest- Tag: X- Asterisk- Guest- Uniqueid: Alert- info: Content- Type: application/sdp Content- Length: 242 v=0 o=root IN IP4 s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=ptime:20 a=sendrecv GUEST ß PBX SIP/ Trying Via:SIP/2.0/UDP :5060;branch=z9hG4bK546a1def;received=;rport=5060 From: " " <sip: @>;tag=as1dddca7a To: <sip: @ > Call- ID: 490e49cf f0007e5ce47d80dd1@ CSeq: 102 INVITE

35 Contact: GUEST ß PBX SIP/ OK Via:SIP/2.0/UDP :5060;branch=z9hG4bK546a1def;received=;rport=5060 From: " " To: Call- ID: CSeq: 102 INVITE Contact: Content- Type: application/sdp Content- Length: 220 v=0 o=root IN IP s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off GUEST à PBX ACK sip: @ SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK3afc8626;rport From: " " <sip: @>;tag=as1dddca7a To: <sip: @ >;tag=as577af7ce Contact: <sip: @> Call- ID: 490e49cf f0007e5ce47d80dd1@ CSeq: 102 ACK Max- Forwards: 70

36 GUEST ß PBX BYE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK5b3130a7;rport From: To: " " Call- ID: CSeq: 102 BYE Max- Forwards: GUEST à PBX SIP/ OK Via:SIP/2.0/UDP :5060;branch=z9hG4bK5b3130a7;received= ;rport=5060 From: To: " " Call- ID: CSeq: 102 BYE Supported: replaces Contact:

37 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 user IP Address> Examples of SIP messages are as follows: PBX USER 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 GUEST à PBX INVITE sip: @ SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK0b7fb7b8;rport From: " " <sip: @>;tag=as0f1a5f0c To: <sip: @ > Contact: <sip: @> Call- ID: 1aa4d60711e0817d731834f474d958b0@ CSeq: 102 INVITE

38 Max- Forwards: 70 Date: Fri, 09 Jul :27:46 GMT Supported: replaces X- Asterisk- Guest- Tag: X- Asterisk- Guest- Uniqueid: Alert- info: Content- Type: application/sdp Content- Length: 242 v=0 o=root IN IP4 s=session c=in IP t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=ptime:20 a=sendrecv PBX à GUEST SIP/ Trying Via: SIP/2.0/UDP :5060;branch=z9hG4bK0b7fb7b8;received=;rport=5060 From: " " <sip: @>;tag=as0f1a5f0c To: <sip: @ > Call- ID: 1aa4d60711e0817d731834f474d958b0@ CSeq: 102 INVITE Contact: <sip: @ > PBX à GUEST SIP/ Busy Here

39 Via: SIP/2.0/UDP :5060;branch=z9hG4bK0b7fb7b8;received=;rport=5060 From: " " To: Call- ID: CSeq: 102 INVITE Contact: GUEST à PBX Transmitting (NAT) to GUEST ACK sip: SIP/2.0 Via: SIP/2.0/UDP :5060;branch= z9hg4bk0b7fb7b8;rport From: " " To: Contact: Call- ID: CSeq: 102 ACK Max- Forwards: 70