agile sip trunk ip-pbx connection manual (asterisk) · 3.1 sip message when you register the...

AGILE SIP TRUNK IP-PBX Connection Manual

(Asterisk)

1. Login to CID (Customer ID) Login https://manager.agile.ne.jp/login.php

2. Go to SIP

USERNAME

Password

3. BUY SIP TRUNK

SIP

SIP List

Buy SIP Trunk

SIP Trunk

Termination

List of SIP TRUNK

Buy SIP Trunk

UID (SIP trunk)

Additional channel SIP trunk Quantity

• Purchase SIP TRUNK

• Add Quantity: UID (SIP TRUNK) = 1 Additional Channel SIP TRUNK = 1

• ADD to CART

• Next

• Next

• Purchase

4. Go to SIP TRUNK LIST

SIP TRUNK LIST LIST OF SIP TRUNK

UID NAME UID

NAM

Channel (Number of Simultaneous call)

Default: 2 Channels for Incoming &

Outgoing

NEXT: PURCHASE DID

5.

PHONE LIST:

• Phone list

• Buy / Purchase Phone Number (DID)

• Cancellation Phone Number

• Disturb

• Transmission Regulation

Choose Buy / Purchase Phone Number (DID)

Phone List

CLICK THIS

BUY PHONE NUMBER (Choose Provider (KDDI, NTT) and search Number base on Area code

AREA

CODE

SEARCH

PICK

NEXT /

SEND

Go back to DID LIST (Phone LIST)

(The DID you purchase is listed here) *Now you can configure AgilePhone for SIP Trunk

Note: UID can be use with multiple DID

Ex. UID DID

OOOO22138 => 0345131495 0368302379 0671763839

DID NUMBER LIST

UID Associated with SIP

Update

Block Diagram of the Inbound and Outbound To: <sip:[email protected]> From: “agile networks” Alert-info number of destination is set <sip:[email protected]>;tag=as5dd4ea>

645 646

Of "SIP message" when sendingIncoming DID is set in the

“To” Header

Of "SIP message" when sending Set the caller ID to "From header"

� CONFIGURATION EXAMPLE

1. Configuration Examples account in Asterisk:

UID : 0000221328

Password : “Your password”

DID Destination : 0345131495, 0368302739 Caller ID : 0368302739, 0345131495 Two cases of agile SIP trunk and SIP extension (645-646)

· DID destination: the case of "0345131495" is to arrive at the "645" of the extension number. · DID destination: the case of "0368302739" is to arrive at the "646" of the extension number. • When you call from "645" to outgoing caller ID to be set to "0,345,131,495". • When you call from "646" to outgoing caller ID to be set to "0,368,302,739". -------------- sip.conf -------------- [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp register => 0000221328:password@siptr [siptr] type=friend username=0000221328 secret=password context=inbound canreinvite=no host=voip3017.agile.ne.jp insecure=port,invite disallow=all allow=ulaw Continue………

[200] type=friend username=645 secret=645pass host=dynamic context=outbound-1 [201] type=friend username=646 secret=646pass host=dynamic context=outbound-2 ------------------ extensions.conf ------------------ [general] writeprotect=no priorityjumping=yes [inbound] ;exten => Destination DID, 1,Dial(SIP/EXTENSION,120,t) ;exten => Destination DID, 2,Congestion ;exten => Destination DID,102,Busy exten => 0345131495, 1,Dial(SIP/645,120,t) exten => 0345131495, 2,Congestion exten => 0345131495,102,Busy exten => 0368302739, 1,Dial(SIP/646,120,t) exten => 0368302739, 2,Congestion exten => 0368302739,102,Busy ;[outbound] ;exten => _0., 1,Set(CALLERID(num)=Caller ID) ;exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) ;exten => _0., 3,Congestion ;exten => _0.,103,Busy

[outbound-1] exten => _ XXX, 1,Set(CALLERID(num)= 0345131495) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing Extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301 exten => _0., 1,Set(CALLERID(num)= 0345131495) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Congestion exten => _0.,104,Busy [outbound-2] exten => _ XXX, 1,Set(CALLERID(num)= 0368302739) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing Extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301 exten => _0., 1,Set(CALLERID(num)= 0368302739) exten => _0., 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _0., 3,Congestion exten => _0.,104,Busy

2. Configuration example to limit the number of simultaneous calls for each group in

Asterisk

• Group 1: ”Limit 2” number of simultaneous calls Extensions: 201~202, Phone Number: 0345131495

• Group 2: “Limit 3” number of simultaneous calls Extensions: 301~302, Phone Number: 0344368713

• UID agile server registered in the guest: 0000221328

• Login server (guest server agile): Voip3017.agile.ne.jp

-------------- sip.conf -------------- [general] allowguest=no maxexpirey=3600 defaultexpirey=3600 context=extd port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw language=jp

register=>0000221328:[email protected]/0000221328 [0000221328] type=friend username=0000221328 secret=password host= voip3017.agile.ne.jp context=inbound ; One Extension Group [201] type=friend context=group1_outbound username=201 secret=password host=dynamic [202] type=friend context=group1_outbound username=202 secret=password host=dynamic ; Two Extension Group [301] type=friend context=group2_outbound username=301 secret=password host=dynamic [302] type=friend context=group2_outbound username=302 secret=password host=dynamic

-------------- extensions.conf -------------- [general] writeprotect=no priorityjumping=yes ; An example of channel limit (incoming) [inbound] ; Group 1 exten => 0345131495, 1,NoOp(EXTEN: ${EXTEN}) exten => 0345131495, 2,Set(GROUP(CALLS)=GROUP1) exten => 0345131495, 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0345131495, 4,Set(MAXCALLS=2) exten => 0345131495, 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] | Hangup) exten => 0345131495, 6,Dial(SIP/201&SIP/202,120) exten => 0345131495, 7,Congestion exten => 0345131495,106,Busy ; Group 2 exten => 0344368713, 1,NoOp(EXTEN: ${EXTEN}) exten => 0344368713, 2,Set(GROUP(CALLS)=GROUP1) exten => 0344368713, 3,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => 0344368713, 4,Set(MAXCALLS=3) exten => 0344368713, 5,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] | Hangup) exten => 0344368713, 6,Dial(SIP/301&SIP/302,120) exten => 0344368713, 7,Congestion exten => 0344368713,106,Busy ; An example of channel limit (outbound) ; Group 1 [group1_outbound] exten => _ XXX, 1,Set(CALLERID(num)= 0345131495) exten => _ XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) This rule is for dialing Extension number. exten => _ XXX,, 3,Congestion _XXX means 3 digit any number. exten => _ XXX,,104,Busy ex. 200, 201, 640, 301 exten => _0., 1,Set(CALLERID(num)= 0345131495) exten => _0., 2,Set(CALLERID(name)=GROUP1) exten => _0., 3,Set(GROUP(CALLS)=GROUP1) exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP1@CALLS)}) exten => _0., 5,Set(MAXCALLS=2) exten => _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] | Hangup) exten => _0., 7,Dial(SIP/${EXTEN}@0000221328,120) exten => _0., 8,Congestion exten => _0.,106,Busy

; Group 2 [group2_outbound] exten => _XXX, 1,Set(CALLERID(num)= 0344368713) exten => _XXX, 2,Dial(SIP/${EXTEN}@siptr,120,T) exten => _ XXX, 3,Congestion exten => _ XXX ,104,Busy exten => _0., 1,Set(CALLERID(num)= 0344368713) exten => _0., 2,Set(CALLERID(name)=GROUP2) exten => _0., 3,Set(GROUP(CALLS)=GROUP2) exten => _0., 4,Set(CURRENTCALLS=${GROUP_COUNT(GROUP2@CALLS)}) exten => _0., 5,Set(MAXCALLS=3) exten => _0., 6,ExecIf($[${CURRENTCALLS} > ${MAXCALLS}] | Hangup) exten => _0., 7,Dial(SIP/${EXTEN}@0000221328,120) exten => _0., 8,Congestion exten => _0.,106,Busy

3. Technical Data

3.1 SIP message when you register the user's information to the guest PBX server:

� Authenticates the user's PBX to the guest server, register the address information and information UID.

Examples of SIP messages as follows:

PBX USER 1.2.1.1

Guest Server 113.34.235.106

Agile UID Sign up to the guest server Guest Server IP Address

6: SIP message of the user’s information when you register to PBX Guest server.

3.1.1. PBX �� GUEST

REGISTER sip:113.34.235.106 SIP/2.0

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4e9b3e05;rport

From: <sip: [email protected]>;tag=as04bc6a95

To: <sip: [email protected]>

Call-ID: [email protected]

CSeq: 1749 REGISTER

User-Agent: Asterisk PBX

Max-Forwards: 70

Expires: 120

Contact: <sip: [email protected]>

Event: registration

Content-Length: 0

3.1.2. GUEST �� PBX

SIP/2.0 100 Trying

Via:SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4e9b3e05;received=1.2.1.1;rport=5060




CSeq: 1749 REGISTER


Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces


Content-Length: 0

3.1.3. GUEST�� PBX

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4e9b3e05;received=1.2.1.1;rport=5060


To: <sip: [email protected]>;tag=as245298a3


CSeq: 1749 REGISTER



Supported: replaces

WWW-Authenticate: Digest algorithm=MD5, realm="voip3024.agile.ne.jp", nonce="3deff552"

Content-Length: 0


REGISTER sip: 113.34.235.106 SIP/2.0

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;rport

From: <sip: [email protected]>;tag=as2031f6e2



CSeq: 1750 REGISTER


Max-Forwards: 70

Authorization: Digest username="0000185475", realm="voip3024.agile.ne.jp", algorithm=MD5,

uri="sip: 113.34.235.106", nonce="3deff552", response="bace343abbe8362868dba84e58d7e056",

opaque=""

Expires: 120


Event: registration

Content-Length: 0


SIP/2.0 100 Trying

Via:SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;received=1.2.1.1;rport=5060




CSeq: 1750 REGISTER



Supported: replaces


Content-Length: 0


SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1db71efa;received=1.2.1.1;rport=5060


To: <sip: [email protected]>;tag=as245298a3


CSeq: 1750 REGISTER



Supported: replaces

Expires: 120

Contact: <sip: [email protected]>;expires=120

Date: Mon, 05 Jul 2010 04:20:13 GMT

Content-Length: 0

3.2. When calling from the user to the guest server PBX:

� PBX user set caller ID from header.

� From header Name field value can be set freely.

� From: "name" <sip: Caller ID@Guest Server IP Domain Name＞

� Examples of SIP messages as follows:

PBX USER 1.2.1.1


Guest Server IP Address

Display Name is Set Free

Caller ID

Callee

Start the Conversation

To end the call

7: Outgoing SIP message from PBX user Guest Server


INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;rport

From: "agile networks" <sip:[email protected]>;tag=as5dd4eaee

To: <sip:[email protected]>

Contact: <sip:[email protected]>


CSeq: 102 INVITE


Max-Forwards: 70

Date: Fri, 02 Jul 2010 03:05:26 GMT


Content-Type: application/sdp

Content-Length: 267

v=0

o=root 22702 22702 IN IP4 1.2.1.1

s=session

c=IN IP4 1.2.1.1

t=0 0

m=audio 18572 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK17bf4505;received=1.2.1.1;rport=5060

From: " agile networks " <sip: [email protected]>;tag=as5dd4eaee

To: <sip:[email protected]>;tag=as4abe0e65


CSeq: 102 INVITE



Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm="voip3024.agile.ne.jp", nonce="23a44cfd"

Content-Length: 0


ACK sip:[email protected] SIP/2.0



To: <sip:[email protected]>;tag=as4abe0e65



CSeq: 102 ACK


Max-Forwards: 70

Content-Length: 0



Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;rport





CSeq: 103 INVITE


Max-Forwards: 70

Proxy-Authorization: Digest username="0000185475", realm="voip3024.agile.ne.jp",

algorithm=MD5, uri="sip:[email protected]", nonce="23a44cfd",

response="cc6c5a668cbd435dee31c767981ff710", opaque=""

Date: Fri, 02 Jul 2010 03:05:26 GMT



Content-Length: 267

v=0

o=root 22702 22703 IN IP4 1.2.1.1

s=session

c=IN IP4 1.2.1.1

t=0 0

m=audio 18572 RTP/AVP 0 8 3 101



a=rtpmap:3 GSM/8000


a=fmtp:101 0-16



SIP/2.0 100 Trying

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK4fc267d7;received=1.2.1.1;rport=5060




CSeq: 103 INVITE



Supported: replaces


Content-Length: 0


SIP/2.0 180 Ringing



To: <sip:[email protected]>;tag=as54380085


CSeq: 103 INVITE



Supported: replaces


Content-Length: 0


SIP/2.0 183 Session Progress





CSeq: 103 INVITE



Supported: replaces



Content-Length: 242

v=0

o=root 4414 4414 IN IP4 113.34.235.106

s=session

c=IN IP4 113.34.235.106

t=0 0

m=audio 18922 RTP/AVP 0 101



a=fmtp:101 0-16


a=ptime:20

a=sendrecv


SIP/2.0 200 OK





CSeq: 103 INVITE



Supported: replaces



Content-Length: 242

v=0

o=root 4414 4415 IN IP4 113.34.235.106

s=session

c=IN IP4 113.34.235.106

t=0 0




a=fmtp:101 0-16


a=ptime:20

a=sendrecv



Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK6c101c7f;rport





CSeq: 103 ACK


Max-Forwards: 70

Content-Length: 0


BYE sip:[email protected] SIP/2.0


From: <sip:[email protected]>;tag=as54380085

To: "agile networks" <sip:[email protected]>;tag=as5dd4eaee


CSeq: 102 BYE


Max-Forwards: 70

Content-Length: 0


SIP/2.0 200 OK

Via:SIP/2.0/UDP

113.34.235.106:5060;branch=z9hG4bK166bf514;received=113.34.235.106;rport=5060

From: <sip:[email protected]>;tag=as54380085

To: "agile networks" <sip:[email protected]>;tag=as5dd4eaee


CSeq: 102 BYE




Content-Length: 0

X-Asterisk-HangupCause: Normal Clearing

3.2 PBX User in case the destination was busy when making calls SIP message:

� If originating from the user when the PBX, the destination was busy, from the guest

server 486 Busy Here message is sent to the user PBX.

� Examples of SIP messages originating from the user at the time when the PBX, the

destination was busy.

PBX USER 1.2.1.1



Caller ID

Destination

8: Destination was busy, SIP message originated from PBX user.



Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport

From: "agile networks" <sip:[email protected]>;tag=as48ac6d56




CSeq: 102 INVITE


Max-Forwards: 70

Date: Tue, 06 Jul 2010 10:09:37 GMT



Content-Length: 267

v=0

o=root 22702 22702 IN IP4 1.2.1.1

s=session

c=IN IP4 1.2.1.1

t=0 0

m=audio 14646 RTP/AVP 0 8 3 101



a=rtpmap:3 GSM/8000


a=fmtp:101 0-16



SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;received=1.2.1.1;rport=5060

To: <sip:[email protected]>;tag=as291aca90


CSeq: 102 INVITE



Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm="voip3024.agile.ne.jp", nonce="15a6e863"

Content-Length: 0

3.3.3. PBX �� Guest


Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK63c44c39;rport


To: <sip:[email protected]>;tag=as291aca90



CSeq: 102 ACK


Max-Forwards: 70

Content-Length: 0

3.3.4. PBX ��GUEST


Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport





CSeq: 103 INVITE


Max-Forwards: 70

Proxy-Authorization: Digest username="0000185475", realm="voip3024.agile.ne.jp",

algorithm=MD5, uri="sip:[email protected]", nonce="15a6e863",

response="54ebd3bdb5bab4b621f55fbd3ffe5e0b", opaque=""

Date: Tue, 06 Jul 2010 10:09:37 GMT



Content-Length: 267

v=0

o=root 22702 22703 IN IP4 1.2.1.1

s=session

c=IN IP4 1.2.1.1

t=0 0

m=audio 14646 RTP/AVP 0 8 3 101



a=rtpmap:3 GSM/8000


a=fmtp:101 0-16



SIP/2.0 100 Trying

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;received=1.2.1.1;rport=5060




CSeq: 103 INVITE


low: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces


Content-Length: 0


SIP/2.0 486 Busy Here

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;received=1.2.1.1;rport=5060


To: <sip:[email protected]>;tag=as715c3c5e


CSeq: 103 INVITE



Content-Length: 0



Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK1c6e5fcc;rport


To: <sip:[email protected]>;tag=as715c3c5e



CSeq: 103 ACK


Max-Forwards: 70

Content-Length: 0

3.4 When coming from the guest PBX server to the user:

� Guest server is set to Alert-info header and the To header destination phone number.

To: <sip: Destination phone number@PBX user IP Address>

� Examples of SIP messages as follows:

9: Incoming SIP messages to PBX server from the guest user


PBX USER 1.2.1.1

Caller ID


Destination Guest Server IP Address

IP Address PBX

Start the Conversation

To end call


Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK546a1def;rport

From: "08058913782" <sip:[email protected]>;tag=as1dddca7a




CSeq: 102 INVITE


Max-Forwards: 70

Date: Fri, 02 Jul 2010 05:41:33 GMT


Supported: replaces

X-Asterisk-Guest-Tag: 00008

X-Asterisk-Guest-Uniqueid: 1278049293.36

Alert-info: 0345900938


Content-Length: 242

v=0

o=root 4414 4414 IN IP4 113.34.235.106

s=session

c=IN IP4 113.34.235.106

t=0 0




a=fmtp:101 0-16


a=ptime:20

a=sendrecv

3.4.2. GUEST PBX

SIP/2.0 100 Trying

Via:SIP/2.0/UDP

113.34.235.106:5060;branch=z9hG4bK546a1def;received=113.34.235.106;rport=5060




CSeq: 102 INVITE




Content-Length: 0

3.4.3. GUEST PBX

SIP/2.0 200 OK

Via:SIP/2.0/UDP

13.34.235.106:5060;branch=z9hG4bK546a1def;received=113.34.235.106;rport=5060


To: <sip:[email protected]>;tag=as577af7ce


CSeq: 102 INVITE





Content-Length: 220

v=0

o=root 22702 22702 IN IP4 1.2.1.1

s=session

c=IN IP4 1.2.1.1

t=0 0




a=fmtp:101 0-16




Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK3afc8626;rport


To: <sip:[email protected]>;tag=as577af7ce



CSeq: 102 ACK


Max-Forwards: 70

Content-Length: 0

3.4.5. GUEST PBX

BYE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK5b3130a7;rport

From: <sip:[email protected]>;tag=as577af7ce

To: "08058913782" <sip:[email protected]>;tag=as1dddca7a


CSeq: 102 BYE


Max-Forwards: 70

Content-Length: 0


SIP/2.0 200 OK

Via:SIP/2.0/UDP 1.2.1.1:5060;branch=z9hG4bK5b3130a7;received=1.2.1.1;rport=5060

From: <sip:[email protected]>;tag=as577af7ce

To: "08058913782" <sip:[email protected]>;tag=as1dddca7a


CSeq: 102 BYE



Supported: replaces


Content-Length: 0

3.5 PBX user arrive, the destination was busy SIP message:

� If the extension of the destination terminal was busy all on the part of the user PBX,

PBX from the user

Send a message to the guest server BUSY

� When the user calls to PBX, If the destination was busy An example of the SIP

message as follows:

10: To the user when the user receives PBX, If the destination was busy SIP

message


PBX USER 1.2.1.1

Caller ID


IP Address PBX


Destination


Via: SIP/2.0/UDP 113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;rport

From: "0345900846" <sip:[email protected]>;tag=as0f1a5f0c




CSeq: 102 INVITE


Max-Forwards: 70

Date: Fri, 09 Jul 2010 02:27:46 GMT


Supported: replaces

X-Asterisk-Guest-Tag: 00024

X-Asterisk-Guest-Uniqueid: 1278642466.508

Alert-info: 0345900938


Content-Length: 242

v=0

o=root 4414 4414 IN IP4 113.34.235.106

s=session

c=IN IP4 113.34.235.106

t=0 0




a=fmtp:101 0-16


a=ptime:20

a=sendrecv


SIP/2.0 100 Trying

Via: SIP/2.0/UDP

113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;received=113.34.235.106;rport=5060




CSeq: 102 INVITE




Content-Length: 0


SIP/2.0 486 Busy Here

Via: SIP/2.0/UDP

113.34.235.106:5060;branch=z9hG4bK0b7fb7b8;received=113.34.235.106;rport=5060




CSeq: 102 INVITE


Content-Length: 0


Transmitting (NAT) to GUEST

ACK sip: [email protected] SIP/2.0

Via: SIP/2.0/UDP 113.34.235.106:5060;branch= z9hG4bK0b7fb7b8;rport





CSeq: 102 ACK


Max-Forwards: 70

Content-Length: 0

agile sip trunk ip-pbx connection manual (asterisk) · 3.1 sip message when you register the...

Documents