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SIP Trunk 2 User Manual

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SIP Trunk 2 IP-PBX User Guide
（Asterisk）
Ver1.1.1 2017/11/21

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Summary of Contents for SIP Trunk 2

  • Page 1 SIP Trunk 2 IP-PBX User Guide （Asterisk） Ver1.1.1 2017/11/21...
  • Page 2 Index SIP Trunk 2 Overview ……………………………………………………… Purchase/Settings in Web Portal ……………………………… Configuration Example of your IP-PBX ……………………………… Technical Data ………………………………...

  • Page 3: Sip Trunk 2 Overview

    １．SIP Trunk 2 Overview SIP Trunk 2 is a next generation IP phone service that connects to PBX making an external line call which is compatible to Asterisk, Aspire X IP-PBX. ＜SIP Trunk 2 FEATURE HIGHLIGHTS＞ Compatible to Asterisk, Aspire X PBX.
  • Page 4 DIDs: 0312345678 0312123434 *If SIP trunk2 Unique was purchased BEFORE March 9, 2017, *If SIP trunk2 Unique was purchased BEFORE March 9, 2017, 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.
  • Page 5 Recipient number is set “To header” and “Alert- Caller ID must be set “From header” for outgoing call. Into” in SIP messages for Incoming call. See section 4 ”Technical Data" for more details. See section 4 ”Technical Data" for more details.
  • Page 6 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. Buy additional SIP trunk channel for 2 or more simultaneous external calls. ＜SIP Trunk 2 Purchase Screen>...
  • Page 7 ２．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> ＜Phone Number Purchase Screen> ① ② ① Select “Purchase” at the top menu and choose ”Purchase Phone Number”...
  • Page 8 ① ③ ② 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...
  • Page 9 ③ Unique is used as client user ID of your user PBX end. ④ Item “Name” is where you can name/rename your SIP Trunk account. ⑤ Select authentication method as “Password Authentication” ⑤ Select authentication method as “Password Authentication”...
  • Page 10 ③ Unique is used as client user ID of your user PBX end. ④ Item “Name” is where you can name/rename your SIP Trunk account. ⑤ Select authentication method as “Authentication with IP Address” ⑥ Enter a public IP address / a port number of your IP-PBX *You can add multiple IP addresses/ports from “+Insert”...
  • Page 11 *Please refer p.14 for details ③ Unique is used as client user ID of your user PBX end. ④ Item “Name” is where you can name/rename your SIP Trunk account. ⑤ Select authentication method as “Authentication using Both IP Address and Password”...
  • Page 12 ２．Purchase/Settings in Web Portal ＜SIP Trunk 2 Detailed Settings ・ Channel Reservation Settings> This section briefly discusses sample settings for SIP trunk2 channel reservations: ① Not using channel reservation (default) If Multiple call count = 4, no reservations. This means outbound and inbound channels maximizes all channel limit.

  • Page 13: This Would Mean That The Selected Unique Will Have Allowed Outbound Calls

    ２．Purchase/Settings in Web Portal ＜SIP Trunk 2 Detailed Settings ・ Channel Reservation Settings> ④ Using both outbound and inbound channel reservation ④ Using both outbound and inbound channel reservation If Multiple call count = 4, If Channel reservation for outbound call = 3...

  • Page 14: Purchase/Settings In Web Portal

    Select phone number(s) you desire to assign to SIP Trunk 2 ＜Phone Number List> ＜Phone Number List> ② ② ① 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...
  • Page 15 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 authentication) ; sip.conf (for either password or IP address with password authentication) ; ------------------...
  • Page 16 ３．Configuration Example of your IP-PBX ; ------------------ ; sip.conf (for either password or IP address with password authentication) ; ------------------ [200] [200] type=friend username=200 secret=200pass host=dynamic context=outbound-1 [201] [201] type=friend username=201 secret=201pass host=dynamic context=outbound-2 ;<see also next page for sip.conf for IP address authentication>...
  • Page 17 ３．Configuration Example of your IP-PBX ; ------------------ ; sip.conf (for IP address authentication) ; ------------------ [general] [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=ulaw language=jp [siptr] type=friend context=inbound canreinvite=no host=xxx.xxx.xxx.xxx insecure=port,invite 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...
  • Page 18 ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Congestion exten =>...
  • Page 19 ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. ;prefix 1xx is for special (external) phone numbers such as 117, 177 and so on. exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Congestion exten =>...
  • Page 20 Max multiple count Group ２: Group ２: 301 ～ 302 301 ～ 302 Extensions Extensions Phone Numbers 03-1212-3434 ; ------------------ ; sip.conf (for either password or IP address with password authentication) ; ------------------ [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr=0.0.0.0 bindaddr=0.0.0.0...

  • Page 21: Configuration Example Of Your Ip-Pbx

    ３．Configuration Example of your IP-PBX ; ------------------ ; sip.conf (for either password or IP address with password authentication) ; ------------------ ; Group 1 ; 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...
  • Page 22 ;please add nat=force_rport,comedia instead of nat=yes in case your asterisk is above ver. 11 [peer1] [peer1] type=friend context=inbound Host=xxx.xxx.xxx.xxx nat=yes [peer2] type=friend context=inbound Host=xxx.xxx.xxx.xxx Host=xxx.xxx.xxx.xxx nat=yes ;please refer to P.10 ② for checking host IP address to be configured as peer. ;<see also next page for the rest settings of sip.conf>...
  • Page 23 ３．Configuration Example of your IP-PBX ;-------------- ;sip.conf (IP address authentication) ;-------------- ; Group 1 ; Group 1 [201] type=friend context=group1_outbound username=201 secret=password host=dynamic [202] [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]...
  • Page 24 [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 => 0312345678,6,Dial(SIP/201&SIP/202,120) exten => 0312345678,7,Congestion exten => 0312345678,106,Busy ; Group 2 exten => 0312123434,1,NoOp(EXTEN: ${EXTEN}) exten => 0312123434,2,Set(GROUP(CALLS)=GROUP2) exten =>...
  • Page 25 => _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _1., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}]?Hangup) exten => _1., 7,Dial(SIP/${EXTEN}@0000123456,120) exten => _1., 8,Congestion exten => _0.,106,Busy exten => _ XXX, 1,Dial(SIP/${EXTEN},120,T) exten => _ XXX, 2,Congestion exten => _ XXX, 102,Busy ; Group 2 ;...

  • Page 26: Technical Data

    ■ Sending REGISTER message Is required to register your ID, IP address and port number for authentication. your IP-PBX SIP Trunk 2 xxx.xxx.xxx.xxx 000.000.000.000 Your ID (SIP Trunk 2 unique IP address of number SIP Trunk 2 REGISTER From: <sip: 0000123456@xxx.xxx.xxx.xxx>;tag=as04bc6a95 From: <sip: 0000123456@xxx.xxx.xxx.xxx>;tag=as04bc6a95...
  • Page 27 ４．Technical Data 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: 0000123456@xxx.xxx.xxx.xxx>;tag=as04bc6a95 From: <sip: 0000123456@xxx.xxx.xxx.xxx>;tag=as04bc6a95 To: <sip: 0000123456@xxx.xxx.xxx.xxx> Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1749 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip: 0000123456@000.000.000.000> Contact: <sip: 0000123456@000.000.000.000> Event: registration Content-Length: 0 4.1.2 GUEST →...
  • Page 28 Call-ID: 34d61b985ef9d9c12d819a9c5549471f@127.0.0.1 CSeq: 1750 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="0000123456", realm=" ", algorithm=MD5, xxx.xxx.xxx.xxx uri="sip: xxx.xxx.xxx.xxx", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056", opaque="" Expires: 120 Contact: <sip: 0000123456@000.000.000.000> Event: registration Contact: <sip: 0000123456@000.000.000.000> Event: registration Content-Length: 0 4.1.5 GUEST → PBX SIP/2.0 100 Trying...
  • Page 29 ４．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＞ Phone IP address of SIP SIP Trunk 2...
  • Page 30 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 Authentication Required Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK17bf4505;received=000.000.000.000;rport=5060 From: From: "aiueo "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as4abe0e65...
  • Page 31 Max-Forwards: 70 Content-Length: 0 Content-Length: 0 4.2.4 PBX → GUEST INVITE sip:080AAAAXXXX@xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;rport From: "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx> To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx> Contact: <sip:0312123434@000.000.000.000> Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username=" 0000123456 ", realm="xxx.xxx.xxx.xxx...
  • Page 32 ４．Technical Data 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: From: "aiueo "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx> Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>...

  • Page 33: Technical Data

    ４．Technical Data 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 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as54380085 Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Supported: replaces Contact: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>...
  • Page 34 ４．Technical Data 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 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK4fc267d7;received=000.000.000.000;rport=5060 From: "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as54380085 Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Supported: replaces Contact: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>...
  • Page 35 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK166bf514;rport From: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as54380085 "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as5dd4eaee Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx 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:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as54380085 To: " aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as5dd4eaee Call-ID: Call-ID: 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx 6426c31c421e503b72515b46569f2ee0@xxx.xxx.xxx.xxx...
  • Page 36 ４．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, SIP Trunk 2 IP address of SIP Trunk 2 your IP-PBX CallerID xxx.xxx.xxx.xxx...
  • Page 37 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 Authentication Required Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;received=000.000.000.000;rport=5060 From: " aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as48ac6d56 To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as291aca90 To: <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as291aca90 Call-ID: 1443bb69616709ff719769cc61d28ce0@xxx.xxx.xxx.xxx...
  • Page 38 ４．Technical Data 4.3.3 PBX → GUEST sip:0312123434@xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK63c44c39;rport From: From: "aiueo "aiueo PBX" <sip:0312123434@000.000.000.000>;tag=as48ac6d56 PBX" <sip:0312123434@000.000.000.000>;tag=as48ac6d56 <sip:080AAAAXXXX@xxx.xxx.xxx.xxx >;tag=as291aca90 Contact: <sip:0312123434@000.000.000.000> Call-ID: 1443bb69616709ff719769cc61d28ce0@xxx.xxx.xxx.xxx CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Content-Length: 0 4.3.4 PBX→GUEST INVITE sip:0312123434@xxx.xxx.xxx.xxx...
  • Page 39 User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:0312123434@xxx.xxx.xxx.xxx> Contact: <sip:0312123434@xxx.xxx.xxx.xxx> 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: " From: " aiueo PBX aiueo PBX " <sip:0312123434@000.000.000.000>;tag=as48ac6d56 "...
  • Page 40 ４．Technical Data 4.4. SIP INVITE message of incoming call from SIP Trunk 2 to your IP-PBX SIP To header will be : <sip:Recipient Phone Number@Your IP PBX IP address＞ *SIP Trunk 2 sets the same recipient phone number to Alert-info header as well.
  • Page 41 - - - - a=ptime:20 a=sendrecv 4.4.2. GUEST←PBX 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:0312345678@000.000.000.000> Call-ID: 490e49cf2141339f0007e5ce47d80dd1@xxx.xxx.xxx.xxx CSeq: 102 INVITE CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0312345678@000.000.000.000>...
  • Page 42 ４．Technical Data 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 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:0312345678@000.000.000.000>;tag=as577af7ce Call-ID: 490e49cf2141339f0007e5ce47d80dd1@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0312345678@000.000.000.000> Contact: <sip:0312345678@000.000.000.000> Content-Type: application/sdp Content-Length: 220 o=root 22702 22702 IN IP4 000.000.000.000...
  • Page 43 ４．Technical Data 4.4.5. GUEST ←PBX sip:080AAAAXXXX@xxx.xxx.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 000.000.000.000:5060;branch=z9hG4bK5b3130a7;rport From: <sip:0312345678@000.000.000.000>;tag=as577af7ce From: <sip:0312345678@000.000.000.000>;tag=as577af7ce To: "080AAAAXXXX" <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as1dddca7a Call-ID: 490e49cf2141339f0007e5ce47d80dd1@xxx.xxx.xxx.xxx 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:0312345678@000.000.000.000>;tag=as577af7ce To: "080AAAAXXXX" <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as1dddca7a...
  • Page 44 ４．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, SIP Trunk 2 your IP-PBX IP address of xxx.xxx.xxx.xxx 000.000.000.000...
  • Page 45 PCMU/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 4.5.2 PBX → GUEST SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: "080AAAAXXXX" <sip:0311112222@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip:0312345678@000.000.000.000> Call-ID: Call-ID: 1aa4d60711e0817d731834f474d958b0@xxx.xxx.xxx.xxx 1aa4d60711e0817d731834f474d958b0@xxx.xxx.xxx.xxx CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0312345678@000.000.000.000>...
  • Page 46 ４．Technical Data 4.5.3. PBX → GUEST SIP/2.0 486 Busy Here Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;received=xxx.xxx.xxx.xxx;rport=5060 xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;received=xxx.xxx.xxx.xxx;rport=5060 From: " 080AAAAXXXX" <sip:080AAAAXXXX@xxx.xxx.xxx.xxx>;tag=as0f1a5f0c To: <sip:0312345678@000.000.000.000> Call-ID: 1aa4d60711e0817d731834f474d958b0@xxx.xxx.xxx.xxx CSeq: 102 INVITE Contact: <sip:0312345678@000.000.000.000> Content-Length: 0 4.5.4. GUEST→ PBX Transmitting (NAT) to GUEST ACK sip: SIP/2.0 0312345678@xxx.xxx.xxx.xxx Via:SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK0b7fb7b8;rport...