3.1.11. SJphone softphone
REQUIREMENTS
This softphone supports both the SIP and H323 protocol. You can download it free of charge from http://www.sjlabs.com. Remember that before trying to configure this softphone, you have to be sure that you have already installed Asterisk server and configured the user in the asterisk conf files. We described this in the previous chapter.
For information how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk
REGISTERING USER
This phone is a little bit different from the ones we previously used. It allows you can dial a user without using asterisk server. (phone to phone). However you have to configure a user on the phone in order to call or to be called.
The phone looks like this: The button which gives you access to the settings is right in the middle of the phone.
I will just change the skin for this one. You can choose it from settings/skins or right mouse click, then options and then skins.
Now lets configure a user. Click the Options button or right mouse click and then options. In the section for user information you can add information about yourself. You can add also your picture.
In Call Options select the IP address that you use.
Then in Profiles choose new to add a new profile.
Now you have to type the name you will use while calling, a filename where your information will be stored and the protocol you will use SIP or H323.
After you click OK choose OK here too. The phone will remain with the default options.
Enter your account name and password. They must be whatever you want.
Then select the user you have just added and then choose Use. And here it is how it looks like the status is in use.
You may then set the audio options. My audio device uses USB sound card so I have changed the speaker and microphone to USB. In the common case these are VIA.
USAGE
Now you have the phone and the user. Now you can use a few different ways to make a call:
1. Dial IP_address
2. dial protocol:IP_address
3. dial extension@Asterisk_IP
4. dial protocol:extension@Asterisk_IP
5. dial extension@Asterisk_IP;user=username
6. dial protocol:extension@Asterisk_IP;user=username
* You can dial user by IP. I have an IP phone with IP 10.10.0.22. If you type : 10.10.0.22 or sip:10.10.0.22 you can dial the phone without using asterisk
* My IP phone has extension 8888 and my Asterisk server has IP 10.3.3.25, so I can make a call by typing this: exten@Asterisk_IP i.e. 8888@10.3.3.25
* And also you can dial like this: sip:8888@10.3.3.25 or also 8888@10.3.3.25;user=sjphone or sip:8888@10.3.3.25;user=sjphone.
This softphone supports both the SIP and H323 protocol. You can download it free of charge from http://www.sjlabs.com. Remember that before trying to configure this softphone, you have to be sure that you have already installed Asterisk server and configured the user in the asterisk conf files. We described this in the previous chapter.
For information how to make the configurations in the Asterisk's configuration files please read our tutorial about the Configuring IP Phones for use with Asterisk
REGISTERING USER
This phone is a little bit different from the ones we previously used. It allows you can dial a user without using asterisk server. (phone to phone). However you have to configure a user on the phone in order to call or to be called.
The phone looks like this: The button which gives you access to the settings is right in the middle of the phone.
I will just change the skin for this one. You can choose it from settings/skins or right mouse click, then options and then skins.
Now lets configure a user. Click the Options button or right mouse click and then options. In the section for user information you can add information about yourself. You can add also your picture.
In Call Options select the IP address that you use.
Then in Profiles choose new to add a new profile.
Now you have to type the name you will use while calling, a filename where your information will be stored and the protocol you will use SIP or H323.
After you click OK choose OK here too. The phone will remain with the default options.
Enter your account name and password. They must be whatever you want.
Then select the user you have just added and then choose Use. And here it is how it looks like the status is in use.
You may then set the audio options. My audio device uses USB sound card so I have changed the speaker and microphone to USB. In the common case these are VIA.
USAGE
Now you have the phone and the user. Now you can use a few different ways to make a call:
1. Dial IP_address
2. dial protocol:IP_address
3. dial extension@Asterisk_IP
4. dial protocol:extension@Asterisk_IP
5. dial extension@Asterisk_IP;user=username
6. dial protocol:extension@Asterisk_IP;user=username
* You can dial user by IP. I have an IP phone with IP 10.10.0.22. If you type : 10.10.0.22 or sip:10.10.0.22 you can dial the phone without using asterisk
* My IP phone has extension 8888 and my Asterisk server has IP 10.3.3.25, so I can make a call by typing this: exten@Asterisk_IP i.e. 8888@10.3.3.25
* And also you can dial like this: sip:8888@10.3.3.25 or also 8888@10.3.3.25;user=sjphone or sip:8888@10.3.3.25;user=sjphone.
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