sip trunk 2 ip-pbx user guide asterisk - agile · 5 2．purchase/settings in web portal for...

SIP Trunk 2 IP-PBX User Guide （Asterisk）

Ver1.0.0 2015/08/01 Ver1.0.3 2015/09/17 Ver1.0.4 2015/10/07 Ver1.0.5 2015/10/15 Ver1.0.6 2015/10/23 Ver1.0.7 2016/01/18

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Index

1. SIP Trunk 2 Overview　　　……………………………………………………… 3

2. Purchase/Settings in Web Portal ……………………………… 5

3. Configuration Example of your IP-PBX ……………………………… 12

4. Technical Data ……………………………… 24

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: 05.01 *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, 2015 221.243.8.194 221.243.8.195

101.110.51.82 101.110.51.83

113.41.163.2 113.41.163.3

１．SIP Trunk 2 Overview

3

Ext. 200 Ext. 201

4

１．SIP Trunk 2 Overview

To:<sip:[email protected]>

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:[email protected]>

Caller ID must be set “From header” for outgoing call. See sec>on 4 ”Technical Data" for more details.

Image 1. Configura>on Diagram of Incoming/Outgoing Calls

xxx.xxx.xxx.xxx SIP Trunk 2

Your IP-PBX

DID: 0312123434 DID: 0312345678

0000.0000.0000.0000

*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******]

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２．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>

①　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

③

①

②

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Purchase phone number here *At least one phone number will be needed for external phone calls through SIP Trunk ＜Phone Number Purchase Screen>

①　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.

①

②

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＜SIP Trunk 2 List＞

①　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

①② ③

0000123456

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＜SIP Trunk 2 Detailed Semngs ・ Password Authen>ca>on＞

①　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.

xxx.xxx.xxx.xxx ①② ③ ④ ⑤ ⑥

0000123456

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＜SIP Trunk 2 Detailed Semngs ・ Authen>ca>on with IP Address＞

①　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 authentication method from “Password Authentication” or “Authentication with IP Address” or “Authentication using both IP Address and Password” ⑤　Enter a public IP address of your IP-PBX ⑥　Enter a public port of your IP-PBX. ⑦　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.

xxx.xxx.xxx.xxx ①② ③ ④ ⑤ ⑥ ⑦

0000123456

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＜SIP Trunk 2 Detailed Settings ・ Authentication using both IP Address and Password>

①　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 authentication method from “Password Authentication” or “Authentication with IP Address” or “Authentication using both IP Address and Password” ⑤　Enter your terminal password is used as client user password of your PBX end. ⑥　Enter a public IP address of your IP-PBX. ⑦　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.

①② ③ ④ ⑤ ⑥ ⑦

xxx.xxx.xxx.xxx

0000123456

11


Select phone number(s) you desire to assign to SIP Trunk 2 ＜Phone Number List>

①　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

②

①

〔0000123456〕

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３．Configuration Example of your IP-PBX

3.1. Configura4on Example in Asterisk [Account Example] Unique: 0000123456 Password: password DIDs: 0312345678 , 0312123434 Extensions: 200, 201 Login Server: xxx.xxx.xxx.xxx ※login the web portal to confirm your login server. [SeMngs Example] Incoming call for 0312345678 is to be arrived at Ext. 200. Incoming call for 0312123434 is to be arrived at Ext. 201. Outgoing call from a phone with Ext. 200 is to be called with CallerID: 0312345678 Outgoing call from a phone with Ext. 201 is to be called with CallerID: 0312123434 ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for either password or IP address with password authen>ca>on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp register => 0000123456:password@siptr [siptr] type=friend username=0000123456 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>

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

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; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for IP address authen>ca>on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp [siptr] type=friend context=inbound canreinvite=no host=xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes [peer1] type=friend context=inbound host=221.243.8.194 nat=yes [peer2] type=friend context=inbound host=221.243.8.195 nat=yes [peer3] type=friend context=inbound host=101.110.51.82 nat=yes [peer4] type=friend context=inbound host=101.110.51.83 nat=yes ;<see also next page for the rest seMngs of sip.conf>

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; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; sip.conf (for IP address authen>ca>on) ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [peer5] type=friend context=inbound host=113.41.163.2 nat=yes [peer6] type=friend context=inbound host=113.41.163.3 nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 [200] type=friend username=200 secret=200pass host=dynamic context=outbound-‐1 [201] type=friend username=201 secret=201pass host=dynamic context=outbound-‐2

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; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; extensions.conf ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] writeprotect=no priorityjumping=yes [inbound] exten => 0312345678,1, Dial(SIP/200,120,t) exten => 0312345678,2,Conges>on exten => 0312345678,102,Busy exten => 0312123434,1, Dial(SIP/201,120,t) exten => 0312123434,2,Conges>on exten => 0312123434,102,Busy [outbound-‐1] exten => _0., 1,Set(CALLERID(num)= 0312345678 exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Conges>on exten => _0.,104,Busy exten => _1., 1,Set(CALLERID(num)= 0312345678 exten => _1., 2,Dial(SIP/${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>

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[outbound-‐2] exten => _0., 1,Set(CALLERID(num)= 0312123434) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Conges>on exten => _0.,104,Busy exten => _1., 1,Set(CALLERID(num)= 0312123434) exten => _1., 2,Dial(SIP/${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.

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Group １: Max multiple count 　2 Extensions 　201 ～ 202 Phone Numbers 　03-1234-5678

Group ２: Max multiple count 　3 Extensions 　301 ～ 302 Phone Numbers 　03-1212-3434

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 ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; 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=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp register=>0000123456:[email protected]/0000123456 [0000123456] type=friend username=0000123456 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>

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

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３．Configuration Example of your IP-PBX ;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ;sip.conf (IP address authen4ca4on) ;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp [siptr] type=friend context=inbound canreinvite=no host= xxx.xxx.xxx.xxx insecure=port,invite disallow=all allow=ulaw qualify=yes nat=yes [peer1] type=friend context=inbound host=221.243.8.194 nat=yes [peer2] type=friend context=inbound host=221.243.8.195 nat=yes [peer3] type=friend context=inbound host=101.110.51.82 nat=yes [peer4] type=friend context=inbound host=101.110.51.83 nat=yes ;<see also next page for the rest seMngs of sip.conf>

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;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ;sip.conf (IP address authen4ca4on) ;-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [peer5] type=friend context=inbound host=113.41.163.2 nat=yes [peer6] type=friend context=inbound host=113.41.163.3 nat=yes ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 ; Group 1 [201] type=friend context=group1_outbound username=201 secret=password host=dynamic [202] type=friend context=group1_outbound username=202 secret=password host=dynamic ; Group 2 [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 Example in your Asterisk> ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ ; extensions.conf ; -‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐-‐ [general] writeprotect=no priorityjumping=yes ; Group 1 [inbound] exten => 0312345678,1,NoOp(EXTEN: ${EXTEN}) exten => 0312345678,2,Set(GROUP(CALLS)=GROUP1) exten => 0312345678,3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0312345678,4,Set(MAXCALLS=2) exten => 0312345678,5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => 0312345678,6,Dial(SIP/201&SIP/202,120) exten => 0312345678,7,Conges>on exten => 0312345678,106,Busy ; Group 2 exten => 0312123434,1,NoOp(EXTEN: ${EXTEN}) exten => 0312123434,2,Set(GROUP(CALLS)=GROUP2) exten => 0312123434,3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => 0312123434,4,Set(MAXCALLS=3) exten => 0312123434,5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => 0312123434,6,Dial(SIP/301&SIP/302,120) exten => 0312123434,7,Conges>on exten => 0312123434,106,Busy ;<see also next page for the rest seMngs of extensions.conf>

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<extensions.conf Example in your Asterisk> ; Group 1 [group1_outbound] exten => _0., 1,Set(CALLERID(num)=0312345678) 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}@0000123456,120) exten => _0., 8,Conges>on exten => _0.,106,Busy exten => _1., 1,Set(CALLERID(num)=0312345678) 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}@0000123456,120) exten => _1., 8,Conges>on exten => _0.,106,Busy exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Conges>on exten => _ XXX, 102,Busy ; Group 2 [group2_outbound] exten => _0., 1,Set(CALLERID(num)= 0312123434) exten => _0., 2,Set(CALLERID(name)=GROUP2) exten => _0., 3,Set(GROUP(CALLS)=GROUP2) exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => _0., 5,Set(MAXCALLS=3) exten => _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _0., 7,Dial(SIP/${EXTEN}@0000123456,120) exten => _0., 8,Conges>on exten => _0.,106,Busy exten => _1., 1,Set(CALLERID(num)= 0312123434) 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}@0000123456,120) 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

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４．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.

figure 4.1 SIP flow for REGISTER

※Sending REGISTER message is NOT required in case your authen4ca4on method is “Authen4ca4on with IP Address”

REGISTER From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-ID: [email protected]

your IP-PBX

000.000.000.000 SIP Trunk 2

xxx.xxx.xxx.xxx

1 100 Trying From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]> Call-ID: [email protected]

2 401 Unauthorized From: <sip: [email protected]>;tag=as04bc6a95 To: <sip: [email protected]>;tag=as245298a3 Call-ID: [email protected]

3 REGISTER(with credential information) From: <sip: [email protected]>;tag=as2031f6e2 To: <sip: [email protected]> Call-ID: [email protected]

4 SIP/2.0 100 Trying From: <sip: [email protected]>;tag=as2031f6e2 To: <sip: [email protected]> Call-ID: [email protected]

5 200 OK From: <sip: [email protected]>;tag=as2031f6e2 To: <sip: [email protected]>;tag=as245298a3 Call-ID: [email protected]

6

Your ID (SIP Trunk 2 unique number

IP address of SIP Trunk 2

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4.1.1 PBX → GUEST REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;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: registra>on Content-‐Length: 0 4.1.2 GUEST →　PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;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 4.1.3 GUEST → PBX SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4e9b3e05;received=000.000.000.000;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-‐Authen>cate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx", nonce="3deff552" Content-‐Length: 0

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4.1.4 PBX → GUEST REGISTER sip: xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;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 Authoriza>on: Digest username="0000123456", realm=" xxx.xxx.xxx.xxx ", algorithm=MD5, uri="sip: xxx.xxx.xxx.xxx", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056", opaque="" Expires: 120 Contact: <sip: [email protected]> Event: registra>on Content-‐Length: 0 4.1.5 GUEST → PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;received=000.000.000.000;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 4.1.6 GUEST → PBX SIP/2.0 200 OK Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1db71efa;received=000.000.000.000;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


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＞

INVITE From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-ID: [email protected]

407 Proxy Authentication Required From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as4abe0e65 Call-ID: [email protected]

INVITE(with credential information) From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]> Call-ID: [email protected]

180 Ringing From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

183 Session Progress From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

200 OK From: "aiueo PBX" <[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee To: <sip:[email protected]>;tag=as54380085 Call-ID: [email protected]

BYE From: <sip:[email protected]>;tag=as54380085 To: "aiueo PBX" <[email protected]>;tag=as5dd4eaee Call-ID: [email protected]

200 OK From: <sip:[email protected]>;tag=as54380085 To: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee Call-ID: [email protected]

SIP Trunk 2 xxx.xxx.xxx.xxx

your IP-PBX 000.000.000.000

Phone Display Name CallerID

IP address of SIP Trunk 2 server

starting a call

Terminating a call

1

2

3

4

5

6

7

8

9

10

11

Receiver Phone

Number

28


4.2.1 PBX → GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rport From: "aiueo PBX" <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: applica>on/sdp Content-‐Length: 267 v=0 o=root 22702 22702 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 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 -‐ -‐ -‐ -‐ 4.2.2　GUEST → PBX SIP/2.0 407 Proxy Authen>ca>on Required Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;received=000.000.000.000;rport=5060 From: "aiueo PBX" <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-‐Authen>cate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="23a44cfd" Content-‐Length: 0

29


4.2.3 PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;rport From: "aiueo PBX" <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 4.2.4 PBX → GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;rport From: "aiueo PBX" <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-‐Authoriza>on: Digest username=" 0000123456 ", realm="xxx.xxx.xxx.xxx ", 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: applica>on/sdp Content-‐Length: 267 v=0 o=root 22702 22703 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 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 -‐ -‐ -‐ -‐

30


4.2.5 GUEST → PBX SIP/2.0 100 Trying Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <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 4.2.6. GUEST → PBX SIP/2.0 180 Ringing Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <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

31


4.2.7 GUEST → PBX SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <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: applica>on/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx 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=p>me:20 a=sendrecv

32


4.2.8 GUEST → PBX SIP/2.0 200 OK Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <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: applica>on/sdp Content-‐Length: 242 v=0 o=root 4414 4415 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx 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=p>me:20 a=sendrecv 4.2.9 PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK6c101c7f;rport From: " aiueo PBX " <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

33


4.2.10 GUEST → PBX BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;rport From: <sip:[email protected]>;tag=as54380085 To: "aiueo PBX" <sip:[email protected]>;tag=as5dd4eaee Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 4.2.11. PBX → GUEST SIP/2.0 200 OK Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;received=xxx.xxx.xxx.xxx;rport=5060 From: <sip:[email protected]>;tag=as54380085 To: " aiueo PBX " <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

34


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,

figure 4.3 SIP flow including Busy message while outgoing call


your IP-PBX 000.000.000.000 CallerID


1

2

3

4

5

6

7

INVITE From:　"aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-ID: [email protected]

407 Proxy Authentication Required From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as291aca90 Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as291aca90 Call-ID: [email protected]

INVITE(with authentication information) From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]> Call-ID: [email protected]

SIP/2.0 486 Busy Here From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Call-ID: [email protected]

ACK From: "aiueo PBX" <sip:[email protected]>;tag=as48ac6d56 To: <sip:[email protected]>;tag=as715c3c5e Call-ID: [email protected]

35


4.3.1　　PBX → GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rport From: "aiueo PBX" <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: applica>on/sdp Content-‐Length: 267 v=0 o=root 22702 22702 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 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 -‐ -‐ -‐ -‐ 4.3.2　　GUEST→ PBX SIP/2.0 407 Proxy Authen>ca>on Required Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 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-‐Authen>cate: Digest algorithm=MD5, realm="xxx.xxx.xxx.xxx ", nonce="15a6e863" Content-‐Length: 0

36


4.3.3　　PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rport From: "aiueo PBX" <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 4.3.4　　PBX→GUEST INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rport From: " aiueo PBX " <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-‐Authoriza>on: Digest username="0000123456", realm="xxx.xxx.xxx.xxx ", algorithm=MD5, uri="sip:[email protected] ", nonce="15a6e863", response="54ebd3bdb5bab4b621f55�d3ffe5e0b", opaque="" Date: Tue, 06 Jul 2010 10:09:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-‐Type: applica>on/sdp Content-‐Length: 267 v=0 o=root 22702 22703 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 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 -‐ -‐ -‐ -‐

37


4.3.5　　GUEST→ PBX SIP/2.0 100 Trying Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:[email protected]>;tag=as48ac6d56 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 4.3.6. GUEST → PBX SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;received=000.000.000.000;rport=5060 From: " aiueo PBX " <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 4.3.7 PBX → GUEST ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK1c6e5fcc;rport From: " aiueo PBX " <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

38


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.

figure 4.4 SIP INVITE flow (incoming)


your IP-PBX 000.000.000.000

IP address of your IP-PBX

1

2

3

4

5

6

CallerID

INVITE From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]> Call-ID: [email protected]

200 OK From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Call-ID: [email protected]

ACK From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip:[email protected]>;tag=as577af7ce Call-ID: [email protected]

BYE From: <sip:[email protected]>;tag=as577af7ce To: “ 080AAAAXXXX " <sip:[email protected]>;tag=as1dddca7a Call-ID: [email protected]

200 OK From: <sip:[email protected]>;tag=as577af7ce To: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a Call-ID: [email protected]

Recipient


Starting a call

Terminating a call

39


4.4.1　　 GUEST→PBX INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;rport From: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a To: <sip: 0312345678 @000.000.000.000> 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: 0312345678 Content-‐Type: applica>on/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx 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=p>me:20 a=sendrecv 4.4.2. GUEST←PBX SIP/2.0 100 Trying Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip: 080AAAAXXXX @xxx.xxx.xxx.xxx>;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

40


4.4.3. GUEST ←PBX SIP/2.0 200 OK Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK546a1def;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <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: applica>on/sdp Content-‐Length: 220 v=0 o=root 22702 22702 IN IP4 000.000.000.000 s=session c=IN IP4 000.000.000.000 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 -‐ -‐ -‐ -‐ 4.4.4 GUEST →PBX ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3afc8626;rport From: "080AAAAXXXX" <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

41


4.4.5. GUEST ←PBX BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;rport From: <sip:[email protected]>;tag=as577af7ce To: "080AAAAXXXX" <sip:[email protected]>;tag=as1dddca7a Call-‐ID: [email protected] CSeq: 102 BYE User-‐Agent: Asterisk PBX Max-‐Forwards: 70 Content-‐Length: 0 4.4.6. GUEST →PBX SIP/2.0 200 OK Via:SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;received=000.000.000.000;rport=5060 From: <sip:[email protected]>;tag=as577af7ce To: "080AAAAXXXX" <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

42


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,

figure 4.5 SIP flow including Busy message while incoming call


your IP-PBX 000.000.000.000

IP address of SIP Trunk 2

server

1

2

3

4

CallerID

INVITE From: "080AAAAXXXX" <sip:080AAAAXXXX"@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

100 Trying From: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

486 Busy Here From: "080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

ACK From: " 080AAAAXXXX" " <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-ID: [email protected]

Recipient IP address of your IP-PBX

43


4.5.1　　 GUEST → PBX INVITE sip:[email protected] SIP/2.0 Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;rport From:" 080AAAAXXXX"<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: 0312345678 Content-‐Type: applica>on/sdp Content-‐Length: 242 v=0 o=root 4414 4414 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx 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=p>me:20 a=sendrecv 4.5.2　　PBX → GUEST SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <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

44


4.5.3.　　PBX → GUEST SIP/2.0 486 Busy Here Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: " 080AAAAXXXX" <sip:[email protected]>;tag=as0f1a5f0c To: <sip:[email protected]> Call-‐ID: [email protected] CSeq: 102 INVITE Contact: <sip:[email protected]> Content-‐Length: 0 4.5.4.　　GUEST→ PBX Transmimng (NAT) to GUEST ACK sip: [email protected] SIP/2.0 Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7�7b8;rport From:"080AAAAXXXX"<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

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