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Customer has a G3v12 system. Calls to a VDN are randomly experiencing one-way audio. This did not occur until 2 weeks ago. No changes have been made to VDN, vector, hunt group, or agents. Each agent is using a digital 6408D phone. The only thing I see is an event during some of the calls...

Download FreePbx. Good support and community for assistance. Freepbx.org

Quazimotto, I would look at FreePbx.org. Similar to Elastix, but still in development. If you are set on Elastix, look at Issabel. Elastix was purchased by 3cx. My understanding is the Issabel is a fork off of Elastix since the purchase. Both FreePbx and Issabel are based on Asterisk, so...

Yes, easily run on these. More simultaneous conversations = more ram needed, but 2gb of ram will work.

Are you sure you are using the correct serial port?

Understand about no phones connected to Asterisk. I would set one up for testing, dialing the message on / off digits. Could even set up softphone on your pc. My gut tells me Definity requires a station to turn the lights on with the code, not a trunk.

An additional thought, are message waiting lights even necessary? Are you thinking too traditionally? Now days, people want the voicemail emailed to them as an attachment. They no longer want to play the voicemail from their phone anyway. If you are getting the call from Avaya and putting it...

I would set up a single channel off of the T1 to look like a 24 single stations on the Definity side. Connect to FreePbx as 24 separate trunks so that you can, if need be select which of the 24 trunks, you send the calls out. Connect a sip phone to Freepbx, send 1062235 out over a select...

So dialing 1062235 from a Freepbx phone turns a light on? or from Avaya phone?

With Avaya Audix, it used a CLan interface and analog ports. Would it be best to put a network hub in between a working interface like this, use something like wireshark to capture the interactions and then use a CLan interface instead of T1/PRI?

Additional question, instead of setting up as a trunk, would it work better to setup as 24 separate stations? Should still get the callerid from phone system. I am not sure Definity will take MWI feature code from a trunk. Years ago I setup a call center off of a Definity with Asterisk in...

Whitelist your ip address. I would use chansip. Most vanilla if you know what I mean. Check you sip and chansip settings, ports. most sip phones want to use 5060.

Interested. Let me know how I can contribute.

Great idea. Have thought the same, with the cost of voicemail systems going crazy, this is awesome. Is there a way to insert a pause in your digit string? Like 106,,,2234. If you manually dial the mwi codes from an analog phone, is it code [tone] extension? Have you giving console...

Fail2ban blocking the ip of the phone?

I have to assume that the sip settings are not correct. Server and or port. user and password registration is not hitting, so something more basic.

Have you tried going into Asterisk CLI in verbose mode? asterisk -rvvvvvvv Watch from CLI when the phone tries to register.

From Asterisk CLI, do you show the sip trunk as registered? Looks like wrong password.

Customer just changed carriers. New PRI. All is working. On incoming calls they are now getting name and telephone number, but display is filled with caller's name. If they have the choice, they would rather just see the caller's telephone number. Is there a way to see all of the information...

My guess is the outbound callerid. If it is sending a callerid that the pri does not like, it will reject. You will probably have to force the outbound callerid.