SipTone 2 is a harphone produced by ipDialog. You can find information also on the phone’s site http://www.ipdialog.com/products.htm. Here we will have a look at the phone’s menu and functions as well as using it to call throuhg Asterisk.
To connect the phone plug one end of an Ethernet cable to the PC input and the other end to your PC LAN card. Plug the free end of your Internet cable to the phone’s ‘Net’ input. The other end must be plugged where your Internet is coming from. Plug the adapter in its jack and connect it to the electricity network.
ADJUSTING
Now when you have connected the phone to the network you can add an user. Here is how the telephone looks like.
You can adjust the phone from the phone menu (to access it use the MENU/HANGUP Key) and also from the phone’s site in your local area network. We will use the site to adjust the phone because it is easier and more user friendly.
To see the IP address of your phone just click the INFO/MUTE Key of your phone. The second line you see on the phone’s display is the IP you look for. Now just type http://<Phone_IP_Address> and a dialog window prompting for username and password will appear.
Note: If your browser cannot open the site maybe the IP address is not in your local network. In this case just go to the phone’s menu choose ‘3)Settings’/’2)Phone Settings’/’2)Config Phone’/’3)Network’/’1)DHCP’/’1)Use DHCP’. So now when you reboot the telephone it will have an IP from your local network. Type again the IP address in an internet browser and in the dialog window log with user:admin and password:admin. Below is the menu of the site.
1. General
This menu gives you general information for the phone like version of the phone, serial number and uptime.
2. Network Setup
In the dropdown field choose MANUAL and type the IP address you want for the telephone in the first text field (make sure the IP is not used by anybody). I choose 10.10.0.99 as you see. Note that after you change the IP address of the phone the site where you can access the setting will also be different. Gateway must be the IP address of your gateway (see you network settings). You also have to provide one or more IP for the DNS server that you use.
3. Phone Configuration
Here I register an user which in a while I will also add to Asterisk. You have to add some ‘Full Name’ and ‘User ID’. It does not really matter what they are but it is prefferable to have some connection with the user using the phone. Dial plan field has to be filled with the number which will dial the phone. This number of course must be a valid extension in extensions.conf on Asterisk.
4. Servers
In REGISTAR choose MANUAL and in SIP URL add your Asterisk server in the following format: protocol (the phone supports just SIP), user, Asterisk IP Address – sip:user@Asterisk_IP. In my case you see it is sip:ivan_new@10.3.3.25.
Remember: user ivan_new has to be added to sip.conf in Asterisk. Expire time is the time after which if nobody picks up the call will expire (here the default value is 3600sec which is one hour you). Server password must be the secret value of the user registered to Asterisk in sip.conf
5. Phonebook
In phonebook you can add contacts and keep them stored together.
6. Change Password
You can change the password which you use to log in. The dafault one is admin. Make sure you commit the change otherwise it will not have effect.
7. Advanced
Here you can make advanced changes to the phone. The ports for the SIP ans RTP are set by default as well as the Audio Codec (it is set to G.711 u-Law).
SETTING ASTERISK
To have the phone in work you have to add the user and extension you are going to use (or you already registered to the phone) in sip.conf and extensions.conf
1.sip.conf
Above is sip.conf and as you see I create user ivan_new of type friend (can call and can be called) with username ivan_new and the same password. I set the host IP to dynamic and add the user to the tutorial context.
2.extennions.conf
Here I just add the extension I declared in ‘Phone Configuration’ to dial the phone. As you see it is number 6789 which dials ivan_new who is actualy registered on the phone.
exten => 6789,1,Dial(SIP/ivan_new)
For more information about how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk
User Comments
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Jinho (skan1603 at empal dot com) 05 January 2006 12:33:09 We use two phone.
One is X-Lite. the Other is ipDialog.
When we make call from ipDialog to X-Lite. Call is Terminated, and ipDialog become Lockup state.
Please help me.
reference my sip debug log
*CLI>
*CLI>
<-- SIP read from 192.168.1.189:5060:
REGISTER sip:192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK72CF2788D8C34F68BFC2F25941114E73
From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Contact: "test" <sip:test@192.168.1.189:5060>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61434 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0
--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.189 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK72CF2788D8C34F68BFC2F25941114E73;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61434 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:test@192.168.1.180>
Content-Length: 0
---
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK72CF2788D8C34F68BFC2F25941114E73;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>;tag=as3406f1af
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61434 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:test@192.168.1.180>
WWW-Authenticate: Digest realm="asterisk", nonce="66ca7905"
Content-Length: 0
---
Scheduling destruction of call 'CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180' in 15000 ms
<-- SIP read from 192.168.1.189:5060:
REGISTER sip:192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK74F5F4C11D3C4710AF1E45949ECF6A18
From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Contact: "test" <sip:test@192.168.1.189:5060>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61435 REGISTER
Expires: 1800
Authorization: Digest username="test",realm="asterisk",nonce="66ca7905",response="f2aca0cc1962620bd3247387242ea1da",uri="sip:192.168.1.180"
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.189 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK74F5F4C11D3C4710AF1E45949ECF6A18;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61435 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:test@192.168.1.180>
Content-Length: 0
---
-- Registered SIP 'test' at 192.168.1.189 port 5060 expires 1800
-- Saved useragent "X-Lite release 1105x" for peer test
Transmitting (no NAT) to 192.168.1.189:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.189:5060;rport;branch=z9hG4bK74F5F4C11D3C4710AF1E45949ECF6A18;received=192.168.1.189From: test <sip:test@192.168.1.180>;tag=2843828580
To: test <sip:test@192.168.1.180>;tag=as3406f1af
Call-ID: CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180
CSeq: 61435 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 1800
Contact: <sip:test@192.168.1.189:5060>;expires=1800
Date: Thu, 05 Jan 2006 11:11:38 GMT
Content-Length: 0
---
Scheduling destruction of call 'CD07CB9580D44B679473C1AD4F9115B3@192.168.1.180' in 15000 ms
<-- SIP read from 192.168.1.183:3072:
REGISTER sip:192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 00007878-f0f08888@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 26111 REGISTER
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00002a7f-f0f0da8f
Supported: timer
To: "ivan_new" <sip:ivan_new@192.168.1.180>
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Expires: 3600
Content-Length: 0
--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.183 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00002a7f-f0f0da8f
To: "ivan_new" <sip:ivan_new@192.168.1.180>
Call-ID: 00007878-f0f08888@192.168.1.183
CSeq: 26111 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:ivan_new@192.168.1.180>
Content-Length: 0
--- (13 headers 16 lines)---
Using INVITE request as basis request - 000029b2-f0f0d942@192.168.1.183
Sending to 192.168.1.183 : 5060 (non-NAT)
Found user 'ivan_new'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.183:5014
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing),
combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4321 in tutorial (domain 192.168.1.180)
list_route: hop: <sip:ivan_new@192.168.1.183>
Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28969 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4321@192.168.1.180>
Content-Length: 0
---
-- Executing Dial("SIP/ivan_new-caac", "SIP/test") in new stack
We're at 192.168.1.180 port 12938
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 192.168.1.189:5060:
INVITE sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK57c33663;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 05 Jan 2006 11:12:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263
--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.189:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:test@192.168.1.189:5060>
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Transmitting (no NAT) to 192.168.1.189:5060:
ACK sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK3edf6a0c;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/test-73e2 answered SIP/ivan_new-caac
We're at 192.168.1.180 port 17270
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.183:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.183:5060;received=192.168.1.183
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 28969 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:4321@192.168.1.180>
Content-Type: application/sdp
Content-Length: 263
--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.189:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Transmitting (no NAT) to 192.168.1.189:5060:
ACK sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK77326121;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<-- SIP read from 192.168.1.183:3075:
ACK sip:4321@192.168.1.180 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.183:5060
Call-ID: 000029b2-f0f0d942@192.168.1.183
Contact: "ivan_new" <sip:ivan_new@192.168.1.183>
CSeq: 28969 ACK
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
To: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
User-Agent: ipDialog SipTone 1.2.0 rc V UA
Content-Length: 0
--- (9 headers 0 lines)---
set_destination: Parsing <sip:ivan_new@192.168.1.183> for address/port to send to
set_destination: set destination to 192.168.1.183, port 5060
We're at 192.168.1.180 port 17270
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 14 lines
Reliably Transmitting (no NAT) to 192.168.1.183:5060:
INVITE sip:ivan_new@192.168.1.183 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK6cf0e90e;rport
From: "4321" <sip:4321@192.168.1.180>;tag=as394f0611
To: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=00006b82-f0f09b72
Contact: <sip:4321@192.168.1.180>
Call-ID: 000029b2-f0f0d942@192.168.1.183
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 317
--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.189:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Transmitting (no NAT) to 192.168.1.189:5060:
ACK sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK459d3856;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
set_destination: Parsing <sip:test@192.168.1.189:5060> for address/port to send to
set_destination: set destination to 192.168.1.189, port 5060
Reliably Transmitting (no NAT) to 192.168.1.189:5060:
BYE sip:test@192.168.1.189:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK0fd45348;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:ivan_new@192.168.1.180>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<-- SIP read from 192.168.1.189:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK0fd45348;rport
From: "ivan_new" <sip:ivan_new@192.168.1.180>;tag=as5ea72487
To: <sip:test@192.168.1.189:5060>;tag=4021734357
Contact: <sip:test@192.168.1.189:5060>
Call-ID: 00317cf00b0bc54d3503c9e40cd8ac4a@192.168.1.180
CSeq: 105 BYE
Server: X-Lite release 1105x
Content-Length: 0