AGILE SIP TRUNK IP-PBX Connection Manual (Asterisk)

Transcription

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

2 1. Login to CID (Customer ID) Login USERNAME Password 2. Go to SIP

3 List of SIP TRUNK SIP SIP List Buy SIP Trunk SIP Trunk Termination 3. BUY SIP TRUNK UID (SIP trunk) Additional channel SIP trunk Quantity Buy SIP Trunk

4 Purchase SIP TRUNK Add Quantity: UID (SIP TRUNK) = 1 Additional Channel SIP TRUNK = 1 ADD to CART Next Next Purchase 4. Go to SIP TRUNK LIST UID NAM UID NAME SIP TRUNK LIST LIST OF SIP TRUNK Channel (Number of Simultaneous call) Default: 2 Channels for Incoming & Outgoing

5 NEXT: PURCHASE DID 5. Phone List PHONE LIST: Phone list Buy / Purchase Phone Number (DID) Cancellation Phone Number Disturb Transmission Regulation Choose Buy / Purchase Phone Number (DID) CLICK THIS

6 BUY PHONE NUMBER (Choose Provider (KDDI, NTT) and search Number base on Area code AREA CODE SEARCH PICK NEXT / SEND

7 Go back to DID LIST (Phone LIST) Update UID Associated with SIP DID NUMBER LIST (The DID you purchase is listed here) *Now you can configure AgilePhone for SIP Trunk Note: UID can be use with multiple DID Ex. UID DID OOOO22138 =>

8 Block Diagram of the Inbound and Outbound To: Alert-info number of destination is set From: agile networks Of "SIP message" when sendingincoming DID is set in the To Header Of "SIP message" when sending Set the caller ID to "From header"

9 CONFIGURATION EXAMPLE 1. Configuration Examples account in Asterisk: UID : Password : Your password DID Destination : , Caller ID : , Two cases of agile SIP trunk and SIP extension ( ) DID destination: the case of " " is to arrive at the "645" of the extension number. DID destination: the case of " " is to arrive at the "646" of the extension number. When you call from "645" to outgoing caller ID to be set to "0,345,131,495". When you call from "646" to outgoing caller ID to be set to "0,368,302,739" sip.conf [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr= srvlookup=yes disallow=all allow=ulaw language=jp register => :password@siptr [siptr] type=friend username= secret=password context=inbound canreinvite=no host=voip3017.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 => Destination DID, 1,Dial(SIP/EXTENSION,120,t) ;exten => Destination DID, 2,Congestion ;exten => Destination DID,102,Busy exten => , 1,Dial(SIP/645,120,t) exten => , 2,Congestion exten => ,102,Busy exten => , 1,Dial(SIP/646,120,t) exten => , 2,Congestion exten => ,102,Busy ;[outbound] ;exten => _0., 1,Set(CALLERID(num)=Caller ID) ;exten => _0., ;exten => _0., 3,Congestion ;exten => _0.,103,Busy

11 [outbound-1] exten => _ XXX, 1,Set(CALLERID(num)= ) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing Extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301 exten => _0., 1,Set(CALLERID(num)= ) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Congestion exten => _0.,104,Busy [outbound-2] exten => _ XXX, 1,Set(CALLERID(num)= ) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing Extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301 exten => _0., 1,Set(CALLERID(num)= ) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Congestion exten => _0.,104,Busy 2. Configuration example to limit the number of simultaneous calls for each group in Asterisk sip.conf Group 1: Limit 2 number of simultaneous calls Extensions: 201~202, Phone Number: Group 2: Limit 3 number of simultaneous calls Extensions: 301~302, Phone Number: UID agile server registered in the guest: Login server (guest server agile): Voip3017.agile.ne.jp [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr= srvlookup=yes disallow=all allow=ulaw language=jp

12 [ ] type=friend username= secret=password host= voip3017.agile.ne.jp context=inbound ; One Extension Group [201] type=friend context=group1_outbound username=201 secret=password host=dynamic [202] type=friend context=group1_outbound username=202 secret=password host=dynamic ; Two Extension Group [301] type=friend context=group2_outbound username=301 secret=password host=dynamic [302] type=friend context=group2_outbound username=302 secret=password host=dynamic

13 extensions.conf [general] writeprotect=no priorityjumping=yes ; An example of channel limit (incoming) [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 ; 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} > ${MAXCALLS}] Hangup) exten => , 6,Dial(SIP/301&SIP/302,120) exten => , 7,Congestion exten => ,106,Busy ; An example of channel limit (outbound) ; Group 1 [group1_outbound] exten => _ XXX, 1,Set(CALLERID(num)= ) exten => _ XXX, 2,Dial(SIP/${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)= ) 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}@ ,120) exten => _0., 8,Congestion exten => _0.,106,Busy

14 ; 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}@ ,120) exten => _0., 8,Congestion exten => _0.,106,Busy

15 3. Technical Data 3.1 SIP message when you register the user's information to the guest PBX server: Authenticates the user's PBX to the guest server, register the address information and information UID. Examples of SIP messages as follows: PBX USER Guest Server Agile UID Sign up to the guest server Guest Server IP Address 6: SIP message of the user s information when you register to PBX Guest server.

16 PBX 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 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: GUEST 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="voip3024.agile.ne.jp", nonce="3deff552"

17 PBX 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="voip3024.agile.ne.jp", algorithm=md5, uri="sip: ", nonce="3deff552", response="bace343abbe dba84e58d7e056", opaque="" Expires: 120 Contact: <sip: Event: registration GUEST 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:

18 GUEST 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. When calling from the user to the guest server PBX: PBX user set caller ID from header. From header Name field value can be set freely. From: "name" <sip: Caller Server IP Domain Name> Examples of SIP messages as follows:

19 Callee PBX USER Display Name is Set Free Caller ID Guest Server Guest Server IP Address Start the Conversation To end the call 7: Outgoing SIP message from PBX user Guest Server

20 PBX 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 GUEST 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 Proxy-Authenticate: Digest algorithm=md5, realm="voip3024.agile.ne.jp", nonce="23a44cfd" PBX GUEST

21 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 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="voip3024.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 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp: a=silencesupp:off GUEST PBX

22 SIP/ Trying Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "agile networks" To: Call-ID: CSeq: 103 INVITE Supported: replaces Contact: GUEST PBX SIP/ Ringing Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "agile networks" To: Call-ID: CSeq: 103 INVITE Supported: replaces Contact:

23 GUEST 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 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 a=ptime:20 a=sendrecv

24 GUEST 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 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 a=ptime:20 a=sendrecv PBX 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

25 GUEST 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 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

26 3.2 PBX User in case the destination was busy when making calls SIP message: If originating from the user when the PBX, the destination was busy, from the guest server 486 Busy Here message is sent to the user PBX. Examples of SIP messages originating from the user at the time when the PBX, the destination was busy. PBX USER Caller ID Guest Server Destination Guest Server IP Address 8: Destination was busy, SIP message originated from PBX user.

27 PBX 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 GUEST 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="voip3024.agile.ne.jp", nonce="15a6e863"

28 PBX 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 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="voip3024.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 a=rtpmap:101 telephone-event/8000 a=fmtp: a=silencesupp:off

29 GUEST PBX SIP/ Trying Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;received= ;rport=5060 From: "agile networks" To: Call-ID: CSeq: 103 INVITE low: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: GUEST PBX SIP/ Busy Here Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;received= ;rport=5060 From: "agile networks" To: Call-ID: CSeq: 103 INVITE Contact: PBX GUEST ACK SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" To: Contact: Call-ID: CSeq: 103 ACK Max-Forwards: 70

30 3.4 When coming from the guest PBX server to the user: Guest server is set to Alert-info header and the To header destination phone number. To: <sip: Destination phone user IP Address> Examples of SIP messages as follows: PBX USER Caller ID Guest Server Destination Guest Server IP Address IP Address PBX Start the Conversation To end call 9: Incoming SIP messages to PBX server from the guest user GUEST PBX

31 INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK546a1def;rport From: " " To: Contact: Call-ID: 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 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 a=ptime:20 a=sendrecv GUEST 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 Contact: <sip: @ >

32 GUEST 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 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: GUEST PBX

33 BYE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK5b3130a7;rport From: To: " " Call-ID: CSeq: 102 BYE Max-Forwards: GUEST PBX SIP/ OK Via:SIP/2.0/UDP :5060;branch=z9hG4bK5b3130a7;received= ;rport=5060 From: To: " " Call-ID: CSeq: 102 BYE Supported: replaces Contact: 3.5 PBX user arrive, the destination was busy SIP message:

34 If the extension of the destination terminal was busy all on the part of the user PBX, PBX from the user Send a message to the guest server BUSY When the user calls to PBX, If the destination was busy An example of the SIP message as follows: PBX USER Caller ID Guest Server Destination IP Address PBX Guest Server IP Address 10: To the user when the user receives PBX, If the destination was busy SIP message GUEST PBX

35 INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK0b7fb7b8;rport From: " " To: Contact: Call-ID: CSeq: 102 INVITE 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 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 a=ptime:20 a=sendrecv PBX 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: @ >

36 PBX GUEST SIP/ Busy Here Via: SIP/2.0/UDP :5060;branch=z9hG4bK0b7fb7b8;received= ;rport=5060 From: " " To: Call-ID: CSeq: 102 INVITE Contact: GUEST 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