keith
08-15-2008, 11:21 AM
Since H.323 is not a core feature of Asterisk and requires 3rd party libraries that are at unpredictable states all the time, Twisted Pair can't in good faith officially support Asterisk in any version of WAVE below 4.0.
Now that WAVE has SIP support, I can finally talk about Asterisk-WAVE interoperability.
This is a short tutorial in the method I use for Asterisk-WAVE configuration using SIP.
For ease of use, I will focus on the FreePBX frontend, the most commonly available web frontend for Asterisk. You can do this using flat files as well, but I prefer to help people using the frontend because it's much easier and streamlined.
There's two parts of this. First covers Media Server SIP trunking, and the second to enable the Dispatch SIP Dialer and utilize it.
For WAVE Media Server SIP Trunking:
In FreePBX:
Login > General Settings. Scroll to the bottom Select 'Yes' on 'Allow anonymous Inbound SIP Calls?'
Extensions > Create Generic SIP Device > Submit
User Extension: (number you wish to use)
Display Name: WAVE Media Server
secret: anything (not actually used but FreePBX requires it)
Now click submit. You should see 'WAVE Media Server' appear on the right menu in FreePBX. Click on it and scroll to the 'dial' line.
Change the dial line to look like this:
SIP/(ext number)@(WAVE Media Server IP):(port)
Example: SIP/3000@70.103.200.238:5060
If your Dispatch client has internet capabilities, you could test out the line above which is our WAVE Server at TPS. If it works, you'll hear a 'Welcome to WAVE' when dialing 3000 from any Asterisk-connected phones. Then you can try your own Asterisk box.
WAVE will always attempt to resolve the Called number with any Session ID configured. If there's no Session ID that matches the dialed extension you will receive the generic 'Welcome to WAVE, please enter Session ID'.
In WAVE for WAVE Dispatch Dialer functionality:
Interfaces > Registrar is defined.
This should be the IP of the Asterisk machine. Port is usually 5060, unless you changed the default. For FreePBX you want 'Use Username for authentication' to not be checked. FreePBX only allows you to configure the username as the extension name.
At the present moment WAVE requires all extensions to have the same password so keep this in mind when building them in FreePBX.
User > Profiles > IP Telephony is checked and correct SIP Registrar is selected.
Users > Users > username > E164 number(s) are defined. These are the extensions you will build in FreePBX.
In FreePBX:
Extensions > Create Generic SIP Device > Submit
User Extension: (number you wish to use in Dispatch)
Display Name: WAVE Dispatcher - Name of Dispatcher
secret: password for VoIP user. WAVE requires all of these to be the same, so make sure it's the same for all extensions.
Voicemail is optional and beyond the scope of this document.
Now you should be able log into Dispatch and have the dialer functional and make and receive calls to any other Asterisk-connected endpoints.
Now that WAVE has SIP support, I can finally talk about Asterisk-WAVE interoperability.
This is a short tutorial in the method I use for Asterisk-WAVE configuration using SIP.
For ease of use, I will focus on the FreePBX frontend, the most commonly available web frontend for Asterisk. You can do this using flat files as well, but I prefer to help people using the frontend because it's much easier and streamlined.
There's two parts of this. First covers Media Server SIP trunking, and the second to enable the Dispatch SIP Dialer and utilize it.
For WAVE Media Server SIP Trunking:
In FreePBX:
Login > General Settings. Scroll to the bottom Select 'Yes' on 'Allow anonymous Inbound SIP Calls?'
Extensions > Create Generic SIP Device > Submit
User Extension: (number you wish to use)
Display Name: WAVE Media Server
secret: anything (not actually used but FreePBX requires it)
Now click submit. You should see 'WAVE Media Server' appear on the right menu in FreePBX. Click on it and scroll to the 'dial' line.
Change the dial line to look like this:
SIP/(ext number)@(WAVE Media Server IP):(port)
Example: SIP/3000@70.103.200.238:5060
If your Dispatch client has internet capabilities, you could test out the line above which is our WAVE Server at TPS. If it works, you'll hear a 'Welcome to WAVE' when dialing 3000 from any Asterisk-connected phones. Then you can try your own Asterisk box.
WAVE will always attempt to resolve the Called number with any Session ID configured. If there's no Session ID that matches the dialed extension you will receive the generic 'Welcome to WAVE, please enter Session ID'.
In WAVE for WAVE Dispatch Dialer functionality:
Interfaces > Registrar is defined.
This should be the IP of the Asterisk machine. Port is usually 5060, unless you changed the default. For FreePBX you want 'Use Username for authentication' to not be checked. FreePBX only allows you to configure the username as the extension name.
At the present moment WAVE requires all extensions to have the same password so keep this in mind when building them in FreePBX.
User > Profiles > IP Telephony is checked and correct SIP Registrar is selected.
Users > Users > username > E164 number(s) are defined. These are the extensions you will build in FreePBX.
In FreePBX:
Extensions > Create Generic SIP Device > Submit
User Extension: (number you wish to use in Dispatch)
Display Name: WAVE Dispatcher - Name of Dispatcher
secret: password for VoIP user. WAVE requires all of these to be the same, so make sure it's the same for all extensions.
Voicemail is optional and beyond the scope of this document.
Now you should be able log into Dispatch and have the dialer functional and make and receive calls to any other Asterisk-connected endpoints.