Some help to get an MP-112, MP-118, MP-124 working with Asterisk.
In my experience, these devices are robust and reliable, but the GUI configuration process is very convoluted. I'm not sure if that is on purpose or not, but I hope this helps you.
This method DOES work and the attached devices (mostly all Polycom Analog conference room phones) have handled thousands of minutes of voice calls without issue.
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
In my experience, these devices are robust and reliable, but the GUI configuration process is very convoluted. I'm not sure if that is on purpose or not, but I hope this helps you.
This method DOES work and the attached devices (mostly all Polycom Analog conference room phones) have handled thousands of minutes of voice calls without issue.
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
!! USE Internet Explorer for GUI Changes/Updates !!
In asterisk (FreePBX) create a SIP extension and save/submit changes.
Note the extension and secret, you'll need these later. Also, I've encountered situations where the random complex password that FreePBX GUI will generate is too long for the AudioCodes devices, so you may need to reduce the character count.
At time of writing I've really only done this with SIP, I haven't done any PJsip testing yet.
At time of writing I've really only done this with SIP, I haven't done any PJsip testing yet.
---If you have already configured your Audiocodes and just want to enter a new number, down and perform only STEP 4 and STEP 5
STEP 1
Login into your AudioCodes (Admin/Admin is the default user. Note CAP "A")
Login into your AudioCodes (Admin/Admin is the default user. Note CAP "A")
Press SUBMIT
At the top of same window, select "PROXY SET TABLE", button (little arrow)
NOTE: I've found CALLCENTRIC worked better if you select "Using Register"
Press SUBMIT
STEP 3
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.
VoIP -> GW and IP to IP -> Hunt Group -> EndPoint Phone Number
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