Some help to get an MP-112, MP-118, MP-124 working with Asterisk. I'm specifically using the FreePBx distro 13.0.197.22, but I feel that most versions should work.
I've made these work directly with CallCentric as well, the change required are noted in the configuration below. If it will work with them, you can probably make it work with any ITSP
I've made these work directly with CallCentric as well, the change required are noted in the configuration below. If it will work with them, you can probably make it work with any ITSP
Tested on an AudioCodes MP-112 on Firmware 6.60A.342.003
Tested on an AudioCodes MP-118 on Firmware 6.60A.332.002
Tested on an AudioCodes MP-124 on Firmware 6.60A.326.005
Tested on an AudioCodes MP-124 on Firmware 6.60A.326.005
In asterisk (FreePBX) create a SIP extension and save/submit changes. The passwords on the AudioCodes seem to be limited to around 15 characters.
---If you have already configured your Audiocodes and just want to enter a new number, down and perform only STEP 4 and STEP 5
The default ip of these boxes is usually 10.1.10.10 or 10.1.10.11 (model depending). Default user Password is Admin/Admin (It uses a capital "A")
STEP 1
VoIP -> SIP Definitions -> Proxy & Registration
3 Press SUBMIT
4 select "PROXY SET TABLE", button (little arrow)
STEP 2
1 Under "PROXY ADDRESS" put in the IP address of your Asterisk server.
NOTE: I've found CALLCENTRIC worked better if you select "Using Register"
4 Press SUBMIT
STEP 2A
VoIP -> SIP Definitions -> General Parameters
My system used 5061 for SIP (not 5060 PJSIP) so i had to make the following changes. These might not apply to you.
VoIP -> GW and IP to IP -> Analog Gateway -> Caller ID Permission
** This is an important step in the configuration process. The system just doesn't seem to work without doing this step! **
Press SUBMIT
STEP 4
For AudioCodes gateways that are already configured, start at this spot to ADD additional phones to the system.
The Gateway port is the port that the extension is plugged into. Either the little Rj11 jacks on the back, or the 50pin connector on the 24 channel unit.
Enter in the extension (or Callcentric Account) for PHONE NUMBER
- 1 1000 [blank] 0
- 2 1002 [blank] 0
Press SUBMIT (If you are watching asterisk live log, your extensions should show as registered)
STEP 6
VoIP ->GW and IP to IP -> DTMF and Supplementary -> DTMF and Dialing
You should now be able to make some test calls. *43 will get you into an echo test on Asterisk
STEP 7
If all works, select BURN on to save these to memory.
Thanks for this. The AudioCodes UI is pretty complicated and this was exactly what I needed. :)
ReplyDeleteThe interface really isn't that user friendly is it? It took awhile to figure out where they buried the settings and how they work with each other in seemingly random fashion.
Deleteold post byut let me add that if you dont want dialing delays for internal extensions, (ie using it in hospitality settng) then be sure to put some digitmaps in. you place them under digit mapping rules in DTMF and dialing. ie 0|7xxx|*2x would mean if ytou dialed 0 or and 4 digit number starting with 7 or any 3 digit number starting with *2 the call will complete instantly and not wait for inter digit timer.
ReplyDelete