AGILE SIP TRUNK IP-PBX Connection Manual (Asterisk)
1. Login to CID (Customer ID) Login https://manager.agile.ne.jp/login.php USERNAME Password 2. Go to SIP
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
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
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
BUY PHONE NUMBER (Choose Provider (KDDI, NTT) and search Number base on Area code AREA CODE SEARCH PICK NEXT / SEND
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 => 0345131495 0368302379 0671763839
Block Diagram of the Inbound and Outbound To: <sip:0345900938@1.2.1.1> Alert-info number of destination is set From: agile networks <sip:03450001280@113.34.235.106>;tag=as5dd4ea> Of "SIP message" when sendingincoming DID is set in the To Header Of "SIP message" when sending Set the caller ID to "From header" 645 646
CONFIGURATION EXAMPLE 1. Configuration Examples account in Asterisk: UID : 0000221328 Password : Your password DID Destination : 0345131495, 0368302739 Caller ID : 0368302739, 0345131495 Two cases of agile SIP trunk and SIP extension (645-646) DID destination: the case of "0345131495" is to arrive at the "645" of the extension number. DID destination: the case of "0368302739" 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=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp register => 0000221328:password@siptr [siptr] type=friend username=0000221328 secret=password context=inbound canreinvite=no host=voip3017.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 => Destination DID, 1,Dial(SIP/EXTENSION,120,t) ;exten => Destination DID, 2,Congestion ;exten => Destination DID,102,Busy exten => 0345131495, 1,Dial(SIP/645,120,t) exten => 0345131495, 2,Congestion exten => 0345131495,102,Busy exten => 0368302739, 1,Dial(SIP/646,120,t) exten => 0368302739, 2,Congestion exten => 0368302739,102,Busy ;[outbound] ;exten => _0., 1,Set(CALLERID(num)=Caller ID) ;exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) ;exten => _0., 3,Congestion ;exten => _0.,103,Busy
[outbound-1] exten => _ XXX, 1,Set(CALLERID(num)= 0345131495) exten => _ XXX, 2,Dial(SIP/${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,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Congestion exten => _0.,104,Busy [outbound-2] exten => _ XXX, 1,Set(CALLERID(num)= 0368302739) exten => _ XXX, 2,Dial(SIP/${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)= 0368302739) 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: 0345131495 Group 2: Limit 3 number of simultaneous calls Extensions: 301~302, Phone Number: 0344368713 UID agile server registered in the guest: 0000221328 Login server (guest server agile): Voip3017.agile.ne.jp [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=>0000221328:password@voip3017.agile.ne.jp/0000221328 [0000221328] type=friend username=0000221328 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
-------------- extensions.conf -------------- [general] writeprotect=no priorityjumping=yes ; An example of channel limit (incoming) [inbound] ; Group 1 exten => 0345131495, 1,NoOp(EXTEN: ${EXTEN}) exten => 0345131495, 2,Set(GROUP(CALLS)=GROUP1) exten => 0345131495, 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0345131495, 4,Set(MAXCALLS=2) exten => 0345131495, 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] Hangup) exten => 0345131495, 6,Dial(SIP/201&SIP/202,120) exten => 0345131495, 7,Congestion exten => 0345131495,106,Busy ; Group 2 exten => 0344368713, 1,NoOp(EXTEN: ${EXTEN}) exten => 0344368713, 2,Set(GROUP(CALLS)=GROUP1) exten => 0344368713, 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0344368713, 4,Set(MAXCALLS=3) exten => 0344368713, 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] Hangup) exten => 0344368713, 6,Dial(SIP/301&SIP/302,120) exten => 0344368713, 7,Congestion exten => 0344368713,106,Busy ; An example of channel limit (outbound) ; Group 1 [group1_outbound] exten => _ XXX, 1,Set(CALLERID(num)= 0345131495) exten => _ XXX, 2,Dial(SIP/${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}@0000221328,120) 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}@0000221328,120) exten => _0., 8,Congestion exten => _0.,106,Busy
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 1.2.1.1 Guest Server 113.34.235.106 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.
3.1.1. PBX 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: 0000185475@113.34.235.106>;tag=as04bc6a95 To: <sip: 0000185475@113.34.235.106> Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1749 REGISTER Max-Forwards: 70 Expires: 120 Contact: <sip: 0000185475@1.2.1.1> Event: registration 3.1.2. GUEST 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: 0000185475@113.34.235.106>;tag=as04bc6a95 To: <sip: 0000185475@113.34.235.106> Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1749 REGISTER Supported: replaces Contact: <sip: 0000185475@113.34.235.106> 3.1.3. GUEST 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: 0000185475@113.34.235.106>;tag=as04bc6a95 To: <sip: 0000185475@113.34.235.106>;tag=as245298a3 Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1749 REGISTER Supported: replaces WWW-Authenticate: Digest algorithm=md5, realm="voip3024.agile.ne.jp", nonce="3deff552"
3.1.4. PBX GUEST REGISTER sip: 113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;rport From: <sip: 0000185475@113.34.235.106>;tag=as2031f6e2 To: <sip: 0000185475@113.34.235.106> Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1750 REGISTER Max-Forwards: 70 Authorization: Digest username="0000185475", realm="voip3024.agile.ne.jp", algorithm=md5, uri="sip: 113.34.235.106", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056", opaque="" Expires: 120 Contact: <sip: 0000185475@1.2.1.1> Event: registration 3.1.5. GUEST 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: 0000185475@113.34.235.106>;tag=as2031f6e2 To: <sip: 0000185475@113.34.235.106> Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1750 REGISTER Supported: replaces Contact: <sip: 0000185475@113.34.235.106>
3.1.6. GUEST 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: 0000185475@113.34.235.106>;tag=as2031f6e2 To: <sip: 0000185475@113.34.235.106>;tag=as245298a3 Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1750 REGISTER Supported: replaces Expires: 120 Contact: <sip: 0000185475@1.2.1.1>;expires=120 Date: Mon, 05 Jul 2010 04:20:13 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 ID@Guest Server IP Domain Name> Examples of SIP messages as follows:
Callee PBX USER 1.2.1.1 Display Name is Set Free Caller ID Guest Server 113.34.235.106 Guest Server IP Address Start the Conversation To end the call 7: Outgoing SIP message from PBX user Guest Server
3.2.1. PBX GUEST INVITE sip:08058913782@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106> Contact: <sip:0345001280@1.2.1.1> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 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. GUEST 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@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106>;tag=as4abe0e65 Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 102 INVITE Supported: replaces Proxy-Authenticate: Digest algorithm=md5, realm="voip3024.agile.ne.jp", nonce="23a44cfd" 3.2.3. PBX GUEST
ACK sip:08058913782@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106>;tag=as4abe0e65 Contact: <sip:0345001280@1.2.1.1> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 102 ACK Max-Forwards: 70 3.2.4. PBX GUEST INVITE sip:08058913782@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106> Contact: <sip:0345001280@1.2.1.1> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="0000185475", realm="voip3024.agile.ne.jp", algorithm=md5, uri="sip:08058913782@113.34.235.106", 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 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@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@113.34.235.106> 3.2.6. GUEST 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@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106>;tag=as54380085 Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@113.34.235.106>
3.2.7. GUEST 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@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106>;tag=as54380085 Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@113.34.235.106> Content-Type: application/sdp Content-Length: 242 v=0 o=root 4414 4414 IN IP4 113.34.235.106 s=session c=in IP4 113.34.235.106 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 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@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106>;tag=as54380085 Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 103 INVITE Supported: replaces Contact: <sip:08058913782@113.34.235.106> Content-Type: application/sdp Content-Length: 242 v=0 o=root 4414 4415 IN IP4 113.34.235.106 s=session c=in IP4 113.34.235.106 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 GUEST ACK sip:08058913782@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK6c101c7f;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as5dd4eaee To: <sip:08058913782@113.34.235.106>;tag=as54380085 Contact: <sip:0345001280@1.2.1.1> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 103 ACK Max-Forwards: 70
3.2.10. GUEST PBX BYE sip:0345001280@1.2.1.1 SIP/2.0 Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK166bf514;rport From: <sip:08058913782@113.34.235.106>;tag=as54380085 To: "agile networks" <sip:0345001280@113.34.235.106>;tag=as5dd4eaee Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 102 BYE Max-Forwards: 70 3.2.11. PBX GUEST SIP/2.0 200 OK Via:SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK166bf514;received=113.34.235.106;rport=5060 From: <sip:08058913782@113.34.235.106>;tag=as54380085 To: "agile networks" <sip:0345001280@113.34.235.106>;tag=as5dd4eaee Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 102 BYE Contact: <sip:0345001280@1.2.1.1> X-Asterisk-HangupCause: Normal Clearing
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 1.2.1.1 Caller ID Guest Server 113.34.235.106 Destination Guest Server IP Address 8: Destination was busy, SIP message originated from PBX user.
3.3.1. PBX GUEST INVITE sip:0345001028@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as48ac6d56 To: <sip:0345001028@113.34.235.106> Contact: <sip:0345001280@1.2.1.1> Call-ID: 1443bb69616709ff719769cc61d28ce0@113.34.235.106 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. GUEST 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@113.34.235.106>;tag=as291aca90 Call-ID: 1443bb69616709ff719769cc61d28ce0@113.34.235.106 CSeq: 102 INVITE Supported: replaces Proxy-Authenticate: Digest algorithm=md5, realm="voip3024.agile.ne.jp", nonce="15a6e863"
3.3.3. PBX Guest ACK sip:0345001028@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as48ac6d56 To: <sip:0345001028@113.34.235.106>;tag=as291aca90 Contact: <sip:0345001280@1.2.1.1> Call-ID: 1443bb69616709ff719769cc61d28ce0@113.34.235.106 CSeq: 102 ACK Max-Forwards: 70 3.3.4. PBX GUEST INVITE sip:0345001028@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as48ac6d56 To: <sip:0345001028@113.34.235.106> Contact: <sip:0345001280@1.2.1.1> Call-ID: 1443bb69616709ff719769cc61d28ce0@113.34.235.106 CSeq: 103 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="0000185475", realm="voip3024.agile.ne.jp", algorithm=md5, uri="sip:0345001028@113.34.235.106", 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 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@113.34.235.106>;tag=as48ac6d56 To: <sip:0345001028@113.34.235.106> Call-ID: 1443bb69616709ff719769cc61d28ce0@113.34.235.106 CSeq: 103 INVITE low: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:0345001028@113.34.235.106> 3.3.6. GUEST 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@113.34.235.106>;tag=as48ac6d56 To: <sip:0345001028@113.34.235.106>;tag=as715c3c5e Call-ID: 1443bb69616709ff719769cc61d28ce0@113.34.235.106 CSeq: 103 INVITE Contact: <sip:0345001028@113.34.235.106> 3.3.7. PBX GUEST ACK sip:0345001028@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport From: "agile networks" <sip:0345001280@113.34.235.106>;tag=as48ac6d56 To: <sip:0345001028@113.34.235.106>;tag=as715c3c5e Contact: <sip:0345001280@1.2.1.1> Call-ID: 1443bb69616709ff719769cc61d28ce0@113.34.235.106 CSeq: 103 ACK Max-Forwards: 70
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 number@pbx user IP Address> Examples of SIP messages as follows: PBX USER 1.2.1.1 Caller ID Guest Server 113.34.235.106 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 3.4.1. GUEST PBX
INVITE sip:0345900938@1.2.1.1 SIP/2.0 Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK546a1def;rport From: "08058913782" <sip:08058913782@113.34.235.106>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1> Contact: <sip:08058913782@113.34.235.106> Call-ID: 490e49cf2141339f0007e5ce47d80dd1@113.34.235.106 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 113.34.235.106 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 PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK546a1def;received=113.34.235.106;rport=5060 From: "08058913782" <sip:08058913782@113.34.235.106>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1> Call-ID: 490e49cf2141339f0007e5ce47d80dd1@113.34.235.106 CSeq: 102 INVITE Contact: <sip:0345900938@1.2.1.1>
3.4.3. GUEST PBX SIP/2.0 200 OK Via:SIP/2.0/UDP 13.34.235.106:5060;branch=z9hG4bK546a1def;received=113.34.235.106;rport=5060 From: "08058913782" <sip:08058913782@113.34.235.106>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1>;tag=as577af7ce Call-ID: 490e49cf2141339f0007e5ce47d80dd1@113.34.235.106 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 PBX ACK sip:0345900938@1.2.1.1 SIP/2.0 Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK3afc8626;rport From: "08058913782" <sip:08058913782@113.34.235.106>;tag=as1dddca7a To: <sip:0345900938@1.2.1.1>;tag=as577af7ce Contact: <sip:08058913782@113.34.235.106> Call-ID: 490e49cf2141339f0007e5ce47d80dd1@113.34.235.106 CSeq: 102 ACK Max-Forwards: 70 3.4.5. GUEST PBX
BYE sip:08058913782@113.34.235.106 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@113.34.235.106>;tag=as1dddca7a Call-ID: 490e49cf2141339f0007e5ce47d80dd1@113.34.235.106 CSeq: 102 BYE Max-Forwards: 70 3.4.6. GUEST 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@113.34.235.106>;tag=as1dddca7a Call-ID: 490e49cf2141339f0007e5ce47d80dd1@113.34.235.106 CSeq: 102 BYE Supported: replaces Contact: <sip:08058913782@113.34.235.106> 3.5 PBX user arrive, the destination was busy SIP message:
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 1.2.1.1 Caller ID Guest Server 113.34.235.106 Destination IP Address PBX Guest Server IP Address 10: To the user when the user receives PBX, If the destination was busy SIP message 3.5.1. GUEST 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@113.34.235.106>;tag=as0f1a5f0c To: <sip:0345900938@1.2.1.1> Contact: <sip:0345900846@113.34.235.106> Call-ID: 1aa4d60711e0817d731834f474d958b0@113.34.235.106 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 113.34.235.106 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 GUEST SIP/2.0 100 Trying Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;received=113.34.235.106;rport=5060 From: "0345900846" <sip:0345900846@113.34.235.106>;tag=as0f1a5f0c To: <sip:0345900938@1.2.1.1> Call-ID: 1aa4d60711e0817d731834f474d958b0@113.34.235.106 CSeq: 102 INVITE Contact: <sip:0345900938@1.2.1.1>
3.5.3. PBX GUEST SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;received=113.34.235.106;rport=5060 From: "0345900846" <sip:0345900846@113.34.235.106>;tag=as0f1a5f0c To: <sip:0345900938@1.2.1.1> Call-ID: 1aa4d60711e0817d731834f474d958b0@113.34.235.106 CSeq: 102 INVITE Contact: <sip:0345900938@1.2.1.1> 3.5.4. GUEST PBX Transmitting (NAT) to GUEST ACK sip: 0345900938@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 113.34.235.106:5060;branch= z9hg4bk0b7fb7b8;rport From: "0345900846" <sip:0345900846@113.34.235.106>;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