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Transcript of AGILE SIP TRUNK IP-PBX Connection Manual (Asterisk) · 3.1 SIP message when you register the user's...
![Page 1: AGILE SIP TRUNK IP-PBX Connection Manual (Asterisk) · 3.1 SIP message when you register the user's information to the guest PBX server: Authenticates the user's PBX to the guest](https://reader030.fdocuments.in/reader030/viewer/2022021709/5bdceb3809d3f2321d8c3e78/html5/thumbnails/1.jpg)
AGILE SIP TRUNK IP-PBX Connection Manual
(Asterisk)
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1. Login to CID (Customer ID) Login https://manager.agile.ne.jp/login.php
2. Go to SIP
USERNAME
Password
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3. BUY SIP TRUNK
SIP
SIP List
Buy SIP Trunk
SIP Trunk
Termination
List of SIP TRUNK
Buy SIP Trunk
UID (SIP trunk)
Additional channel SIP trunk Quantity
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• 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 &
Outgoing
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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
CLICK THIS
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BUY PHONE NUMBER (Choose Provider (KDDI, NTT) and search Number base on Area code
AREA
CODE
SEARCH
PICK
NEXT /
SEND
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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
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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"
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� 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………
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[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/${EXTEN}@siptr,120,T) ;exten => _0., 3,Congestion ;exten => _0.,103,Busy
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[outbound-1] exten => _ XXX, 1,Set(CALLERID(num)= 0345131495) exten => _ XXX, 2,Dial(SIP/${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
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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
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-------------- 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
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; 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
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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.
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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
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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
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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:
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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
7: Outgoing SIP message from PBX user Guest Server
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3.2.1. PBX ���� GUEST
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport
From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee
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 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: [email protected]>;tag=as5dd4eaee
To: <sip:[email protected]>;tag=as4abe0e65
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="23a44cfd"
Content-Length: 0
3.2.3. PBX ���� GUEST
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ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport
From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee
To: <sip:[email protected]>;tag=as4abe0e65
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
3.2.4. PBX ���� GUEST
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;rport
From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee
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="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
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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
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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
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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
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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
X-Asterisk-HangupCause: Normal Clearing
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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
Guest Server 113.34.235.106
Guest Server IP Address
Caller ID
Destination
8: Destination was busy, SIP message originated from PBX user.
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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
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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 - - - -
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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
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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:
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
To end call
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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]>
![Page 32: AGILE SIP TRUNK IP-PBX Connection Manual (Asterisk) · 3.1 SIP message when you register the user's information to the guest PBX server: Authenticates the user's PBX to the guest](https://reader030.fdocuments.in/reader030/viewer/2022021709/5bdceb3809d3f2321d8c3e78/html5/thumbnails/32.jpg)
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
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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
3.5 PBX user arrive, the destination was busy SIP message:
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� 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
![Page 35: AGILE SIP TRUNK IP-PBX Connection Manual (Asterisk) · 3.1 SIP message when you register the user's information to the guest PBX server: Authenticates the user's PBX to the guest](https://reader030.fdocuments.in/reader030/viewer/2022021709/5bdceb3809d3f2321d8c3e78/html5/thumbnails/35.jpg)
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]>
![Page 36: AGILE SIP TRUNK IP-PBX Connection Manual (Asterisk) · 3.1 SIP message when you register the user's information to the guest PBX server: Authenticates the user's PBX to the guest](https://reader030.fdocuments.in/reader030/viewer/2022021709/5bdceb3809d3f2321d8c3e78/html5/thumbnails/36.jpg)
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