SIP Trunk 2 IP-PBX User Guide Asterisk

Transcription

1 SIP Trunk 2 IP-PBX User Guide Asterisk Ver /08/01 Ver /09/17 Ver /10/07 Ver /10/15 Ver /10/23 Ver /01/18

2 Index 1. SIP Trunk 2 Overview 3 2. Purchase/Settings in Web Portal 5 3. Configuration Example of your IP-PBX Technical Data 24 2

3 SIP Trunk 2 Overview SIP Trunk 2 is a next genera>on IP phone service that connects to PBX making an external line call which is compa>ble to Asterisk, Aspire X IP- PBX. SIP Trunk 2 FEATURE HIGHLIGHTS Compa>ble to Asterisk, Aspire X PBX. Op>ons for Authen>ca>on Method are: Password Authen>ca>on Authen>ca>on with IP Address Authen>ca>on using both IP Address and Password. CPS Call Per Second) has been significantly improved from normal SIP trunk. *Our Cloud PBX Recording Op>on is currently not supported by SIP trunk 2 (If you need the recording op>on, please Contact us) ===== Verified IP- PBX ===== Asterisk Asterisk PBX/1.4.x Asterisk PBX 1.6.x Asterisk PBX 1.8.x Asterisk PBX 11 Asterisk PBX 12 Aspire X IP3WW- 32VOIPDB- A1 version: *IP- PBX versions not listed above are not fully supported by SIP trunk 2. ======================== Please permit on your firewall incoming network traffic from our VoIP server IP addresses with 5060, 10000~20000 UDP ports. Our Server IP address list *as of Oct 23,

4 SIP Trunk 2 Overview SIP Trunk 2 xxx.xxx.xxx.xxx To:<sip: @ > Recipient number is set To header and Alert- Into in SIP messages for Incoming call. See sec>on 4 Technical Data" for more details. From: <sip: @xxx.xxx.xxx.xxx> Caller ID must be set From header for outgoing call. See sec>on 4 Technical Data" for more details. Your IP-PBX DID: DID: Ext. 200 Ext. 201 Image 1. Configura>on Diagram of Incoming/Outgoing Calls *In case of Japanese toll free numbers such as prefix 0120, 0800 and 0570, you should set its background number showing in Phone Number List of the web portal. ex.) A number enclosed in parentheses is its background number. 0120****** [03******] 4

5 Purchase/Settings in Web Portal For purchasing SIP Trunk 2, access the UI of our IP-PBX. Buy additional SIP trunk channel for 2 or more simultaneous external calls. SIP Trunk 2 Purchase Screen> Select Purchase at the top menu and choose Purchase Unique in Circle Management Page 2 Select quantity of SIP trunk 2 3 Click Add to Cart to proceed for your purchase 5

6 Purchase/Settings in Web Portal Purchase phone number here *At least one phone number will be needed for external phone calls through SIP Trunk Phone Number Purchase Screen> Select Purchase at the top menu and choose Purchase Phone Number in Circle Management Page 2 On the Purchase Phone Number page, find your desired phone number by clicking Search button. Add to cart and select Your Cart to proceed. 6

7 Purchase/Settings in Web Portal SIP Trunk 2 List Select SIP Trunk List to open all your SIP trunk account 2 Select the icon under Detail for detailed settings of SIP Trunk (See next page) 3 Your unique is used as client user ID of your user PBX end 7

8 Purchase/Settings in Web Portal SIP Trunk 2 Detailed Semngs Password Authen>ca>on 1 xxx.xxx.xxx.xxx Login server name of SIP Trunk 2 2 Unique is used as client user ID of your user PBX end. 3 Item Name is where you can name/rename your SIP Trunk account. 4 Select your desired authen>ca>on method from Password Authen>ca>on or Authen>ca>on with IP Address or Authen>ca>on using both IP Address and Password 5 Enter your terminal password is used as client user password of your PBX end. 6 Set mul>ple call count. It s 1 by default. Purchase Addi>onal 1 channel for SIP Trunk 2 if you need more than 2 concurrent calls. 8

9 Purchase/Settings in Web Portal SIP Trunk 2 Detailed Semngs Authen>ca>on with IP Address 1 xxx.xxx.xxx.xxx Login server name of SIP Trunk 2 Unique is used as client user ID of your user PBX end. Item Name is where you can name/rename your SIP Trunk account. Select your desired authentication method from Password Authentication or Authentication with IP Address or Authentication using both IP Address and Password Enter a public IP address of your IP-PBX Enter a public port of your IP-PBX. Set multiple call count. It s 1 by default. Purchase Additional 1 channel for SIP Trunk 2 if you need more than 2 concurrent calls. 9

10 Purchase/Settings in Web Portal SIP Trunk 2 Detailed Settings Authentication using both IP Address and Password> xxx.xxx.xxx.xxx Login server name of SIP Trunk 2 Unique is used as client user ID of your user PBX end. Item Name is where you can name/rename your SIP Trunk account. Select your desired authentication method from Password Authentication or Authentication with IP Address or Authentication using both IP Address and Password Enter your terminal password is used as client user password of your PBX end. Enter a public IP address of your IP-PBX. Set multiple call count. It s 1 by default. Purchase Additional 1 channel for SIP Trunk 2 if you need more than 2 concurrent calls. 10

11 Purchase/Settings in Web Portal Select phone number(s) you desire to assign to SIP Trunk 2 Phone Number List> Click Phone Number List to open your Phone Number List. 2 Select SIP Trunk 2 unique for phone number(s) you desire to assign for it 11

12 Configuration Example of your IP-PBX 3.1. Configura4on Example in Asterisk [Account Example] Unique: Password: password DIDs: , Extensions: 200, 201 Login Server: xxx.xxx.xxx.xxx login the web portal to confirm your login server. [SeMngs Example] Incoming call for is to be arrived at Ext Incoming call for is to be arrived at Ext Outgoing call from a phone with Ext. 200 is to be called with CallerID: Outgoing call from a phone with Ext. 201 is to be called with CallerID: ; ; sip.conf (for either password or IP address with password authen>ca>on) ; [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr= srvlookup=yes disallow=all allow=ulaw language=jp register => :password@siptr [siptr] username= secret=password context=inbound canreinvite=no host=xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 ;<see also next page for the rest semngs of sip.conf> 12

13 Configuration Example of your IP-PBX ; ; sip.conf (for either password or IP address with password authen>ca>on) ; [200] username=200 secret=200pass host=dynamic context=outbound- 1 [201] username=201 secret=201pass host=dynamic context=outbound- 2 ;<see also next page for sip.conf for IP address authen4ca4on> 13

14 Configuration Example of your IP-PBX ; ; sip.conf (for IP address authen>ca>on) ; [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr= srvlookup=yes disallow=all allow=ulaw language=jp [siptr] context=inbound canreinvite=no host=xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes [peer1] context=inbound host= nat=yes [peer2] context=inbound host= nat=yes [peer3] context=inbound host= nat=yes [peer4] context=inbound host= nat=yes ;<see also next page for the rest semngs of sip.conf> 14

15 Configuration Example of your IP-PBX ; ; sip.conf (for IP address authen>ca>on) ; [peer5] context=inbound host= nat=yes [peer6] context=inbound host= nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 [200] username=200 secret=200pass host=dynamic context=outbound- 1 [201] username=201 secret=201pass host=dynamic context=outbound- 2 15

16 Configuration Example of your IP-PBX ; ; extensions.conf ; [general] writeprotect=no priorityjumping=yes [inbound] exten => ,1, Dial(SIP/200,120,t) exten => ,2,Conges>on exten => ,102,Busy exten => ,1, Dial(SIP/201,120,t) exten => ,2,Conges>on exten => ,102,Busy [outbound- 1] exten => _0., 1,Set(CALLERID(num)= exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Conges>on exten => _0.,104,Busy exten => _1., 1,Set(CALLERID(num)= exten => _1., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _1., 3,Conges>on exten => _1.,104,Busy ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges>on exten => _ XXX, 102,Busy ; XXX represents 3 digit- extensions. Please adjust digit number as yours. ;<see also next page for the rest semngs of extensions.conf> 16

17 Configuration Example of your IP-PBX [outbound- 2] exten => _0., 1,Set(CALLERID(num)= ) exten => _0., exten => _0., 3,Conges>on exten => _0.,104,Busy exten => _1., 1,Set(CALLERID(num)= ) exten => _1., exten => _1., 3,Conges>on exten => _1.,104,Busy ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges>on exten => _ XXX, 102,Busy ; XXX represents 3 digit- extensions. Please adjust digit number as yours. 17

18 Configuration Example of your IP-PBX 3.2. Configura4on Example to limit mul4ple call count for each extension group in Asterisk. [SeMngs Example] Set max mul>ple call count (for external calls) as 2 for Group 1 Set max mul>ple call count (for external calls) as 3 for Group 2 Group : Group : Max multiple count 2 Extensions Phone Numbers Max multiple count 3 Extensions Phone Numbers ; ; sip.conf (for either password or IP address with password authen>ca>on) ; [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr= srvlookup=yes disallow=all allow=ulaw language=jp register=> :password@xxx.xxx.xxx.xxx/ [ ] username= secret=password host=xxx.xxx.xxx.xxx insecure=port,invite context=inbound qualify=yes nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 ;<see also next page for the rest semngs of sip.conf> 18

19 Configuration Example of your IP-PBX ; ; sip.conf (for either password or IP address with password authen>ca>on) ; ; Group 1 [201] context=group1_outbound username=201 secret=password host=dynamic [202] context=group1_outbound username=202 secret=password host=dynamic ; Group 2 [301] context=group2_outbound username=301 secret=password host=dynamic [302] context=group2_outbound username=302 secret=password host=dynamic ;<see also next page for sip.conf for IP address authen4ca4on> 19

20 Configuration Example of your IP-PBX ; ;sip.conf (IP address authen4ca4on) ; [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr= srvlookup=yes disallow=all allow=ulaw language=jp [siptr] context=inbound canreinvite=no host= xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes [peer1] context=inbound host= nat=yes [peer2] context=inbound host= nat=yes [peer3] context=inbound host= nat=yes [peer4] context=inbound host= nat=yes ;<see also next page for the rest semngs of sip.conf> 20

21 Configuration Example of your IP-PBX ; ;sip.conf (IP address authen4ca4on) ; [peer5] context=inbound host= nat=yes [peer6] context=inbound host= nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 ; Group 1 [201] context=group1_outbound username=201 secret=password host=dynamic [202] context=group1_outbound username=202 secret=password host=dynamic ; Group 2 [301] context=group2_outbound username=301 secret=password host=dynamic [302] context=group2_outbound username=302 secret=password host=dynamic 21

22 Configuration Example of your IP-PBX <extensions.conf Example in your Asterisk> ; ; extensions.conf ; [general] writeprotect=no priorityjumping=yes ; Group 1 [inbound] 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,Conges>on exten => ,106,Busy ; Group 2 exten => ,1,NoOp(EXTEN: ${EXTEN}) exten => ,2,Set(GROUP(CALLS)=GROUP2) exten => ,3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => ,4,Set(MAXCALLS=3) exten => ,5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => ,6,Dial(SIP/301&SIP/302,120) exten => ,7,Conges>on exten => ,106,Busy ;<see also next page for the rest semngs of extensions.conf> 22

23 Configuration Example of your IP-PBX <extensions.conf Example in your Asterisk> ; Group 1 [group1_outbound] 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,Conges>on exten => _0.,106,Busy exten => _1., 1,Set(CALLERID(num)= ) exten => _1., 2,Set(CALLERID(name)=GROUP1) exten => _1., 3,Set(GROUP(CALLS)=GROUP1) exten => _1., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => _1., 5,Set(MAXCALLS=2) exten => _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _1., 7,Dial(SIP/${EXTEN}@ ,120) exten => _1., 8,Conges>on exten => _0.,106,Busy exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges>on exten => _ XXX, 102,Busy ; Group 2 [group2_outbound] exten => _0., 1,Set(CALLERID(num)= ) exten => _0., 2,Set(CALLERID(name)=GROUP2) exten => _0., 3,Set(GROUP(CALLS)=GROUP2) exten => _0., 4,Set(CURRENTCALLS=${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,Conges>on exten => _0.,106,Busy exten => _1., 1,Set(CALLERID(num)= ) exten => _1., 2,Set(CALLERID(name)=GROUP2) exten => _1., 3,Set(GROUP(CALLS)=GROUP2) exten => _1., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => _1., 5,Set(MAXCALLS=3) exten => _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _1., 7,Dial(SIP/${EXTEN}@ ,120) exten => _1., 8,Conges>on exten => _1.,106,Busy exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges>on exten => _ XXX, 102,Busy 23

24 Technical Data 4.1. SIP REGISTER message Sending REGISTER message Is required to register your ID, IP address and port number for authen>ca>on. your IP-PBX Your ID (SIP Trunk 2 unique number IP address of SIP Trunk 2 SIP Trunk 2 xxx.xxx.xxx.xxx REGISTER From: <sip: @xxx.xxx.xxx.xxx>;tag=as04bc6a95 To: <sip: @xxx.xxx.xxx.xxx> Call-ID: 34d61b985ef9d9c12d819a9c f@ Trying From: <sip: @xxx.xxx.xxx.xxx>;tag=as04bc6a95 To: <sip: @xxx.xxx.xxx.xxx> Call-ID: 34d61b985ef9d9c12d819a9c f@ Unauthorized From: <sip: @xxx.xxx.xxx.xxx>;tag=as04bc6a95 To: <sip: @xxx.xxx.xxx.xxx>;tag=as245298a3 Call-ID: 34d61b985ef9d9c12d819a9c f@ REGISTER(with credential information) From: <sip: @xxx.xxx.xxx.xxx>;tag=as2031f6e2 To: <sip: @xxx.xxx.xxx.xxx> Call-ID: 34d61b985ef9d9c12d819a9c f@ SIP/ Trying From: <sip: @xxx.xxx.xxx.xxx>;tag=as2031f6e2 To: <sip: @xxx.xxx.xxx.xxx> Call-ID: 34d61b985ef9d9c12d819a9c f@ OK From: <sip: @xxx.xxx.xxx.xxx>;tag=as2031f6e2 To: <sip: @xxx.xxx.xxx.xxx>;tag=as245298a3 Call-ID: 34d61b985ef9d9c12d819a9c f@ figure 4.1 SIP flow for REGISTER Sending REGISTER message is NOT required in case your authen4ca4on method is Authen4ca4on with IP Address 24

25 Technical Data PBX GUEST REGISTER sip:xxx.xxx.xxx.xxx 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: registra>on GUEST PBX SIP/ Trying Via:SIP/2.0/UDP :5060;branch=z9hG4bK4e9b3e05;received= ;rport=5060 From: <sip: To: <sip: Call- ID: CSeq: 1749 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY 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 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW- Authen>cate: Digest algorithm=md5, realm="xxx.xxx.xxx.xxx", nonce="3deff552" 25

26 Technical Data PBX GUEST REGISTER sip: xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1db71efa;rport From: <sip: >;tag=as2031f6e2 To: <sip: > Call- ID: CSeq: 1750 REGISTER Max- Forwards: 70 Authoriza>on: Digest username=" ", realm=" xxx.xxx.xxx.xxx ", algorithm=md5, uri="sip: xxx.xxx.xxx.xxx", nonce="3deff552", response="bace343abbe dba84e58d7e056", opaque="" Expires: 120 Contact: <sip: Event: registra>on GUEST PBX SIP/ Trying Via:SIP/2.0/UDP :5060;branch=z9hG4bK1db71efa;received= ;rport=5060 From: <sip: >;tag=as2031f6e2 To: <sip: > Call- ID: CSeq: 1750 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip: > GUEST PBX SIP/ OK Via: SIP/2.0/UDP :5060;branch=z9hG4bK1db71efa;received= ;rport=5060 From: <sip: >;tag=as2031f6e2 To: <sip: >;tag=as245298a3 Call- ID: CSeq: 1750 REGISTER Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: <sip: Date: Mon, 05 Jul :20:13 GMT 26

27 Technical Data 4.2. SIP INVITE message of outgoing call from your IP- PBX through SIP Trunk 2 SIP From header should be : From: Phone Display name <sip:callerid@sip Trunk 2 IP address or FQDN your IP-PBX Phone Display Name CallerID IP address of SIP Trunk 2 server SIP Trunk 2 xxx.xxx.xxx.xxx Receiver 1 Phone Number INVITE From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx> Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx 407 Proxy Authentication Required From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as4abe0e65 Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx ACK From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as4abe0e65 Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx INVITE(with credential information) From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx> Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx 100 Trying From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx> Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx 180 Ringing From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx 183 Session Progress From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx 200 OK From: "aiueo PBX" <sip @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx ACK From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx BYE From: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as To: "aiueo PBX" <sip @ >;tag=as5dd4eaee Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx starting a call Terminating a call OK From: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as To: "aiueo PBX" <sip: @ >;tag=as5dd4eaee Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx

28 Technical Data PBX GUEST INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK17bf4505;rport From: "aiueo PBX" To: Contact: Call- ID: CSeq: 102 INVITE Max- Forwards: 70 Date: Fri, 02 Jul :05:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content- Type: applica>on/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 Authen>ca>on Required Via: SIP/2.0/UDP :5060;branch=z9hG4bK17bf4505;received= ;rport=5060 From: "aiueo PBX" <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as4abe0e65 Call- ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy- Authen>cate: Digest algorithm=md5, realm="xxx.xxx.xxx.xxx ", nonce="23a44cfd" 28

29 Technical Data PBX GUEST ACK SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK17bf4505;rport From: "aiueo PBX" 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: "aiueo PBX" To: Contact: Call- ID: CSeq: 103 INVITE Max- Forwards: 70 Proxy- Authoriza>on: Digest username=" ", realm="xxx.xxx.xxx.xxx ", algorithm=md5, nonce="23a44cfd", response="cc6c5a668cbd435dee31c767981ff710", opaque="" Date: Fri, 02 Jul :05:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content- Type: applica>on/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

30 Technical Data GUEST PBX SIP/ Trying Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "aiueo PBX" To: Call- ID: CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: GUEST PBX SIP/ Ringing Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "aiueo PBX" To: Call- ID: CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: 30

31 Technical Data GUEST PBX SIP/ Session Progress Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "aiueo PBX" To: Call- ID: CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content- Type: applica>on/sdp Content- Length: 242 v=0 o=root IN IP4 xxx.xxx.xxx.xxx s=session c=in IP4 xxx.xxx.xxx.xxx t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=p>me:20 a=sendrecv 31

32 Technical Data GUEST PBX SIP/ OK Via: SIP/2.0/UDP :5060;branch=z9hG4bK4fc267d7;received= ;rport=5060 From: "aiueo PBX" To: Call- ID: CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content- Type: applica>on/sdp Content- Length: 242 v=0 o=root IN IP4 xxx.xxx.xxx.xxx s=session c=in IP4 xxx.xxx.xxx.xxx t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=p>me:20 a=sendrecv PBX GUEST ACK sip:080aaaaxxxx@xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK6c101c7f;rport From: " aiueo PBX " <sip: @ >;tag=as5dd4eaee To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as Contact: <sip: @ > Call- ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx CSeq: 103 ACK Max- Forwards: 70 32

33 Technical Data GUEST PBX BYE SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk166bf514;rport From: To: "aiueo PBX" Call- ID: CSeq: 102 BYE Max- Forwards: PBX GUEST SIP/ OK Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk166bf514;received=xxx.xxx.xxx.xxx;rport=5060 From: To: " aiueo PBX " Call- ID: CSeq: 102 BYE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: X- Asterisk- HangupCause: Normal Clearing 33

34 Technical Data 4.3. SIP Busy message while outgoing call in case receiver is on another call Busy message sent by SIP Trunk 2 when receiver is currently on another call, your IP-PBX CallerID IP address of SIP Trunk 2 server SIP Trunk 2 xxx.xxx.xxx.xxx 1 2 INVITE From: "aiueo PBX" <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx> Call-ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx 407 Proxy Authentication Required From: "aiueo PBX" <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as291aca90 Call-ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx ACK From: "aiueo PBX" <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as291aca90 Call-ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx INVITE(with authentication information) From: "aiueo PBX" <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx> Call-ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx 100 Trying From: "aiueo PBX" <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx> Call-ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx SIP/ Busy Here From: "aiueo PBX" <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as715c3c5e Call-ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx ACK From: "aiueo PBX" <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as715c3c5e Call-ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx figure 4.3 SIP flow including Busy message while outgoing call 34

35 Technical Data PBX GUEST INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK63c44c39;rport From: "aiueo PBX" To: Contact: Call- ID: CSeq: 102 INVITE Max- Forwards: 70 Date: Tue, 06 Jul :09:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content- Type: applica>on/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 Authen>ca>on Required Via: SIP/2.0/UDP :5060;branch=z9hG4bK63c44c39;received= ;rport=5060 From: " aiueo PBX " <sip: @ >;tag=as48ac6d56 To: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as291aca90 Call- ID: 1443bb ff719769cc61d28ce0@xxx.xxx.xxx.xxx CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy- Authen>cate: Digest algorithm=md5, realm="xxx.xxx.xxx.xxx ", nonce="15a6e863" 35

36 Technical Data PBX GUEST ACK SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK63c44c39;rport From: "aiueo PBX" To: >;tag=as291aca90 Contact: Call- ID: CSeq: 102 ACK Max- Forwards: PBX GUEST INVITE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;rport From: " aiueo PBX " To: Contact: Call- ID: CSeq: 103 INVITE Max- Forwards: 70 Proxy- Authoriza>on: Digest username=" ", realm="xxx.xxx.xxx.xxx ", algorithm=md5, ", nonce="15a6e863", response="54ebd3bdb5bab4b621f55 d3ffe5e0b", opaque="" Date: Tue, 06 Jul :09:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content- Type: applica>on/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

37 Technical Data GUEST PBX SIP/ Trying Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;received= ;rport=5060 From: " aiueo PBX " To: Call- ID: CSeq: 103 INVITE Allow: 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: " aiueo PBX " To: Call- ID: CSeq: 103 INVITE Contact: PBX GUEST ACK SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK1c6e5fcc;rport From: " aiueo PBX " To: Contact: Call- ID: CSeq: 103 ACK Max- Forwards: 70 37

38 Technical Data 4.4. SIP INVITE message of incoming call from SIP Trunk 2 to your IP- PBX SIP To header will be : To: <sip:recipient Phone Number@Your IP PBX IP address *SIP Trunk 2 sets the same recipient phone number to Alert- info header as well. your IP-PBX Recipient CallerID SIP Trunk 2 xxx.xxx.xxx.xxx INVITE From: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as1dddca7a To: <sip: @ > Call-ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx 100 Trying From: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as1dddca7a To: <sip: @ > Call-ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx 200 OK From: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as1dddca7a To: <sip: @ >;tag=as577af7ce Call-ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx ACK From: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as1dddca7a To: <sip: @ >;tag=as577af7ce Call-ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx BYE From: <sip: @ >;tag=as577af7ce To: 080AAAAXXXX " <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as1dddca7a Call-ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx 200 OK From: <sip: @ >;tag=as577af7ce To: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as1dddca7a Call-ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx IP address of your IP-PBX IP address of SIP Trunk 2 server Starting a call Terminating a call figure 4.4 SIP INVITE flow (incoming) 38

39 Technical Data GUEST PBX INVITE SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk546a1def;rport From: "080AAAAXXXX" To: <sip: Contact: Call- ID: CSeq: 102 INVITE Max- Forwards: 70 Date: Fri, 02 Jul :41:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X- Asterisk- Guest- Tag: X- Asterisk- Guest- Uniqueid: Alert- info: Content- Type: applica>on/sdp Content- Length: 242 v=0 o=root IN IP4 xxx.xxx.xxx.xxx s=session c=in IP4 xxx.xxx.xxx.xxx t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=p>me:20 a=sendrecv GUEST PBX SIP/ Trying Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip: To: <sip: @ > Call- ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip: @ > 39

40 Technical Data GUEST PBX SIP/ OK Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" To: Call- ID: CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content- Type: applica>on/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 xxx.xxx.xxx.xxx:5060;branch=z9hg4bk3afc8626;rport From: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as1dddca7a To: <sip: @ >;tag=as577af7ce Contact: <sip:080aaaaxxxx@xxx.xxx.xxx.xxx> Call- ID: 490e49cf f0007e5ce47d80dd1@xxx.xxx.xxx.xxx CSeq: 102 ACK Max- Forwards: 70 40

41 Technical Data GUEST PBX BYE SIP/2.0 Via: SIP/2.0/UDP :5060;branch=z9hG4bK5b3130a7;rport From: To: "080AAAAXXXX" Call- ID: CSeq: 102 BYE Max- Forwards: GUEST PBX SIP/ OK Via:SIP/2.0/UDP :5060;branch=z9hG4bK5b3130a7;received= ;rport=5060 From: To: "080AAAAXXXX" Call- ID: CSeq: 102 BYE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: 41

42 Technical Data 4.5. SIP Busy message while incoming call in case receiver is on another call Busy message sent by SIP Trunk 2 when receiver is currently on another call, your IP-PBX Recipient IP address of your IP-PBX CallerID SIP Trunk 2 xxx.xxx.xxx.xxx INVITE From: "080AAAAXXXX" <sip:080aaaaxxxx"@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip: @ > Call-ID: 1aa4d60711e0817d731834f474d958b0@xxx.xxx.xxx.xxx 100 Trying From: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip: @ > Call-ID: 1aa4d60711e0817d731834f474d958b0@xxx.xxx.xxx.xxx 486 Busy Here From: "080AAAAXXXX" <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip: @ > Call-ID: 1aa4d60711e0817d731834f474d958b0@xxx.xxx.xxx.xxx ACK From: " 080AAAAXXXX" " <sip:080aaaaxxxx@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip: @ > Call-ID: 6dd7b12f1438e1572cae057f274419e6@ IP address of SIP Trunk 2 server figure 4.5 SIP flow including Busy message while incoming call 42

43 Technical Data GUEST PBX INVITE SIP/2.0 Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk0b7 7b8;rport From:" To: Contact: <sip: Call- ID: CSeq: 102 INVITE Max- Forwards: 70 Date: Fri, 09 Jul :27:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X- Asterisk- Guest- Tag: X- Asterisk- Guest- Uniqueid: Alert- info: Content- Type: applica>on/sdp Content- Length: 242 v=0 o=root IN IP4 xxx.xxx.xxx.xxx s=session c=in IP4 xxx.xxx.xxx.xxx t=0 0 m=audio RTP/AVP a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone- event/8000 a=fmtp: a=silencesupp:off a=p>me:20 a=sendrecv PBX GUEST SIP/ Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk0b7 7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip: @xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip: @ > Call- ID: 1aa4d60711e0817d731834f474d958b0@xxx.xxx.xxx.xxx CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip: @ > 43

44 Technical Data PBX GUEST SIP/ Busy Here Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk0b7 7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: " 080AAAAXXXX" To: Call- ID: CSeq: 102 INVITE Contact: GUEST PBX Transmimng (NAT) to GUEST ACK sip: SIP/2.0 Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hg4bk0b7 7b8;rport To: Contact: Call- ID: CSeq: 102 ACK Max- Forwards: 70 44