AGILE SIP TRUNK
IP-PBX Connection Manual
(Asterisk)
1. Login to CID (Customer ID) Login https://manager.agile.ne.jp/login.php
2. Go to
SIP
USERNAME
Password
3. BUY SIP TRUNK
SIP
SIP List
Buy SIP Trunk
SIP Trunk
Termination
List of SIP TRUNK
Buy SIP Trunk
UID (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
SIP TRUNK LIST
LIST OF SIP TRUNK
UID NAME UID
NAM
Channel (Number of Simultaneous call)
Default:
2 Channels for Incoming &
NEXT:
PURCHASE DID
5.
PHONE LIST:
• Phone list
• Buy / Purchase Phone Number (DID)
• Cancellation Phone Number
• Disturb
• Transmission Regulation
Choose Buy / Purchase Phone Number (DID)
Phone List
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)
(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
DID NUMBER LIST
UID Associated with SIP
Update
Block Diagram of the Inbound and Outbound
To: <sip:[email protected]> From: “agile networks”
Alert-info number of destination is set <sip:[email protected]>;tag=as5dd4ea>
645
646
Of "SIP message" when sendingIncoming DID is set in the
“To” Header
Of "SIP message" when sending Set the caller ID to "From header"
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/${EXTEN}@siptr,120,T) ;exten => _0., 3,Congestion
[outbound-1]
exten => _ XXX, 1,Set(CALLERID(num)= 0345131495)
exten => _ XXX, 2,Dial(SIP/${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
• 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 --- 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=>0000221328:[email protected]/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: [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 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. 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: [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="voip3024.agile.ne.jp", nonce="3deff552" Content-Length: 0
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: [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="0000185475", realm="voip3024.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 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: [email protected]>;tag=as2031f6e2
To: <sip: [email protected]>
Call-ID: [email protected] CSeq: 1750 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces
Contact: <sip: [email protected]> Content-Length: 0
3.1.6. 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: [email protected]>;tag=as2031f6e2
To: <sip: [email protected]>;tag=as245298a3 Call-ID: [email protected] CSeq: 1750 REGISTER
User-Agent: Asterisk PBX
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
Content-Length: 0
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:
PBX USER 1.2.1.1
Guest Server
113.34.235.106
Guest Server IP Address Display Name is Set Free
Caller ID Callee
Start the Conversation
To end the call
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:[email protected]> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 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. 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
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="voip3024.agile.ne.jp", nonce="23a44cfd" Content-Length: 0
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:[email protected]>;tag=as4abe0e65 Contact: <sip:[email protected]> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 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:[email protected]> Contact: <sip:[email protected]> Call-ID: 6426c31c421e503b72515b46569f2ee0@113.34.235.106 CSeq: 103 INVITE User-Agent: Asterisk PBX 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
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 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 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 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 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:[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 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:[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 PBX
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 113.34.235.106:5060;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 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:[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
3.2PBX 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 Guest Server 113.34.235.106 Guest Server IP Address Caller ID Destination
3.3.1. PBX 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. 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:[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="voip3024.agile.ne.jp", nonce="15a6e863" Content-Length: 0
3.3.3. PBX 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 GUEST
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="voip3024.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 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 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 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
3.4When 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:
9: Incoming SIP messages to PBX server from the guest user
3.4.1. GUEST PBX 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
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 113.34.235.106:5060;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 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:[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 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:[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 PBX
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 113.34.235.106:5060;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 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 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
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:
10: To the user when the user receives PBX, If the destination was busy SIP message 3.5.1. GUEST PBX PBX USER 1.2.1.1 Caller ID Guest Server 113.34.235.106 IP Address PBX Guest Server IP Address Destination
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 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 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:[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 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:[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 PBX
Transmitting (NAT) to GUEST
ACK 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 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0
