You don't mention what system you're using - the term Starplus covers a great deal of territory. Nor do you mention the type of lines. Do you have CID enabled on system and from carrier? Do you see CID info when a call rings in? When using analog lines, you must set ring delay timer to at...
Over the past couple of years, we've been installing edge router with SBC and dual WAN ports in order to have SIP fail over. While not inexpensive, they just work, which makes this solution cheaper in the end.
Mike Forrence
www.goteamvox.com
Indeed, if you've had multiple. Check the UDP port and firewall type that each STUN assigns - this may have a bearing. Check Monitor on a failed call, looking at UDP port in particular.
Mike Forrence
www.goteamvox.com
I've actually had a wonky STUN do EXACTLY this! Find another STUN server or (even better) install your own SBC, and trouble will go away.
Mike Forrence
www.goteamvox.com
IP500 controller has two contact closures via an 1/8" phono plug on back of control unit. You'll want to use these to drive separate relays to do the actual unlocking. But it works like a champ.
Mike Forrence
www.goteamvox.com
That's a LOT of red alarms. If it were me, I'd connect up to the serial port on the smartjack and see what internal history shows - it will almost always tell the tale. The menus are fairly easy to navigate. The carrier may not like you connecting up, but it'll help find the trouble. I would...
If this is in your community, it'll likely be faster to simply go to the site and make the changes. How often do you envision changing the config anyway? If you DO anticipate many changes over time, you may be able to install an old PC next to Partner, access the PC via LogMeIn/TeamViewer/etc...
Well, I'd then swap port, as that's all there is. Depending on model/version of the system,you may or may not be able to retain the extension 109 number. There is nothing in config that would cause a phone to randomly place a call without user action, so if that is really what is happening...
Analog trunk/line to FXO gateway to ethernet to wireless bridge end A to wireless bridge end B to Ethernet to FXS gateway to analog station device
Mike Forrence
www.goteamvox.com
In order to keep it simple and robust I'd use a separate audio input on each Telecenter and use a Valcom audio distribution solution to get audio to each site. That allows you to leave current connections alone.
Mike Forrence
www.goteamvox.com
Make a shortcode of # + user's extension, aimed at whatever is appropriate for the situation. i.e. if user is extn 1234 and you want the attempt to result in a busy signal, make #1234 result in barred. That, or tell offending users to not.
Mike Forrence
www.goteamvox.com
Thanks for all of the responses.
@tlpeter - while I don't disagree with you, it's difficult to convince a customer to trash 70+ working telephones because they want a dozen SIP trunks. If it weren't for the departure of ADI, we'd just add SIP licenses and remain at current release, because it...
@Nutritional_Info - I saw that one, but it was mostly about 5X00 series, and just barely mentioned 6400. I was looking for a bit more focused info. Cheers!
Mike Forrence
www.goteamvox.com
HI all,
I have a site that uses many many 6424+ phones and needs to move to PLDS in order to add SIP trunks. While I know that they are no longer supported, does anyone have solid info regarding the latest release where they still function normally? Thanks
Mike Forrence
www.goteamvox.com
Have you looked at Monitor to see if the call arrives in IPO-land? If call does not arrive, it's a network issue. If call does arrive, the Monitor trace will tell you what is unhappy.
Mike Forrence
www.goteamvox.com
A bit depends on how the call is getting to voicemail in the first place.
I would first check Flash 40 CO line voicemail ID. If current value is 102, change it to 100 and it'll just work.
Cheers!
Mike Forrence
www.goteamvox.com
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