Dear posi211,
Well i have no idea at the moment so i will try to find out soon and tell you,
Thank you in advance for your help
Dear Posi211 ,
I found the Model is āPanasonic KX-T123210DBā
Thank you for you help ,
and I tried the IVR and it could not pass me to an other PBX with out using a lot of Digit options that is kind of a Pain.
So I let it be for the time being if any one else has any other Ideas I am very open for discussion 
Thank you
Thatās one of the reasons to program the required digital into the trunk
If you have the Panasonic voicemail you press #8 followed by the extension.
The system is very old and max out could only have 60 phone on it.
Need to install a small freepbx and ditch the old white box. (old yellow box now)
Dear Posi211,
Well when i will do the over Halling i will think about it.
So this PBX is not able to do the trick? Or not?
Thank you in advance
You can add an ATA with FXO ports and become analog devices on the old Panasonic.
Donāt do it.
That system is so old, run away. If the customers not willing to upgrade that old pos then they will never pay you what you are worth.
Dear Posi211,
Yeah thatās they idea is to move forward not backwoodsā¦
Anyway i will see how i will do itā¦
Thank you in advance
Plus, if something works donāt f with it
If it doesnāt work f itā¦
When the current system dies, an upgrade will be inevitable, but for the moment being is okay 
until more stuff start to fail like old phones and etc
Then i will go full voip
Plus does anyone knows how to encrypt all communications from the phone to the PBX?
THANK you
If you donāt have a T1/E1 link between the old pbx and freepbx donāt bother. The cost to connect the two of them through analog extensions and be usefull would be high. Sure you can connect two or three extensions between the two but that is the maximum channels between the two boxes.
If you have freepbx installed and voip phones that support tls and srtp you can encrypt the calls by using these two protocols.
Dear Astbox ,
Yeah thatās very true indeed , but as i mentioned earlier , this is not for production but rather to see there are the limits of the systems , for now I am happy with the set up ,
For instance one of my problems are that I still have VPN problems in regard to the codec , that i still get some part of the conversation lost , and I switched it to TCP instead of UDP it got better but still i think my Internet connection is not up for it. and I even tried a new codec the proprietary G729
and still even worse results that GSM , so For the future I will keep the analogue trunks in place and not more directly to Voip trunks because the internet speed will not suffice.
other problems that i had is that i need to Isolate and encrypt the Voip Communications because anyone with wireshark and 10$ can decrypt all Codecs regardless ā¦ so is like an open book for anyone listening
So that is also an issue to be resolved , but testing that it is resolved ,
Other than that , I love analogue , no delays , no codecs , just 2 conductors and a battery (of sorts)
Thank you in advance for your input ,
Well you donāt have to move from your analog lines. Buy a pstn card to host these lines and switch the pbx with freepbx.
You will be using the same lines but with all the goodies of freepbx. Porting your pbx to voip doesnāt mean that you have to port also your lines.
You keep the same lines and have a sip trunk as a failover or wait until your isp gets you a decent speed connection.
Also with a big box with many fxs on, you can keep your analog phones also and get some sip devices for offices that need extra functionallity.
That is exactly what I am going to do to replace the old PBX , 24 FXS digium card , or maybe 3 of them just for good measure , and a 10 or 8 port FXO card , a Xenon Server with all the bells , and the G729 codec for the ISP ( they use that here )
and some Voip Phones to change the Panasonic Station Proprietary ones that are useless with out the Panasonic PBX
Thank you for your input 
I would suggest you to buy a fxs adapter and have the phones attached to that. Easier to maintain, will offload your system and finally make the cabling easier as usually these units are rack mounted and have all the ports on the front.
Dear astbox this is a different item from the digium or openvox cards?
Is there a Specific brand?
Thank you
Iti is something like this
http://www.openvox.cn/products/voip-gateways/analog-gateways/160/vs-gw1600-analog-series-detail.html
I think digium might have also one and sangoma the same but OpenVox I think is the cheapest.
Also if for any reason the freepbx installation is messed up, with a device like this you can have a warm backup that has also the same number of fxo ports with the same configuration, you simply change the ip and you are up and working much quicker than changing the cards and rerouting all the cables to the new box.
Also this unit from Openvox has five swapable modules, if a port is messed up, you can remove it, plug a new one and you are good to go. There are a bit iffy related to their firmware but overall for the price and versatility are good units. Other devices if a port has a problem, the whole unit must be sent back for a fix, with this you just have a spare module, you replace it and wait for the other one to be fixed.
I am mentioning all this because fxs and fxo ports are prone to failure.
1 Like
Dear Astbox ,
Yeah true that but still it it self id prone to die ( yeah the part that you swap powers itās a good measure ,
but still Iāve seen the online demo , and I am not really satisfied ā¦ is a bit "how you doing ā¦ "
for 350$ and an digium card is several times itās pice for just the card itās kind of saying more than it should ,
but is still an option ,
I will have to look around and see what other people are doing in case of upgrades from this kind of system to a new one and see how they managed it ,
and go from there but As i Said , I didnāt die yet , and If i keep it happy , I think It will out last the phones , and maybe even my self ā¦ who knows ā¦
Things Made in Japan wore awesome Quality stuff, today well idk how are being done.
Also Anyone had any experience with Cisco phone systems ? to be purely Cisco servers, phones , routers and e.t.c ?
I donāt understand what you are saying. You can go with a card, you can go with a big ata. Thatās all I wanted to say.
I didnāt mean that you should do, just mentioning some common problems that might have in mind when choosing hardware. 
Dear Astbox ,
sorry about that , I derailed my self there ā¦
3 things .:
-GUI of the OpenVox is a bit strange ( online demo )
-The feature that you can change cards is nice 
-The gateway can fail too (so regardless if the Asterix server or the gateway still i would have to do something )
I just did a brain fart there and I am sorry
But I am really grateful for your input
I will see what others have done that wore in my situation and go from there and also see what other implementations are done to other customers (of the guy i will hire to do it , or my self )
As I said again , Thank you for your input
Dear All ,
After Making the IVR to Work , but Once I push the button to push me to the trunk to connect me to the other PBX ,
It just drops the line ,
Plus I configured all the incoming routes on both systems to be pointed to the Ring groups and i allows on the peer settings the from-analogue line so that i can accept the call ,and ā
DAHDI Channel DIDsā i configured it for a number so that it can pass it to the other PBX ,
and still drops ,
Anything else i need to configure ?
and any other ideas ?
Thank you in advance
Where in the this whole thread does Dahdi come into your systems, do you even have any Dahdi trunks or extensions ?
Dear Dicko,
Yeah i kind of high jacked the title of the conversation.
But the idea is 3 systems with IAX trunks
And an old Analogue PBX system
One system has an Analogue card with 1 FXO and 1 FXS port
IVR to handle the incoming extension from the analogue PBX.
The idea is to dial from any analog extension of the analogue PBX and get to the IVR then get to any of the IP PBX systems via the IAX trunks
Then i pointed that the IVR will push the call to the trunk of the remote pbx it just drops the line. Disconnects all together. (the call the system does not crash in any way)
So this is the problem currently.
And sometimes i get failed to decrypt packet, or no number or some strange messages on the Log files.
I hope this is sufficient.
Thank you