Not hard at all, you will just need IP phones at remote site and create these on the Panasonic (add a virtual card)
Licenses and DSP card will also be required.
The DSP card is licensed for 4 IP phones out the box but you will need extra if you have more than 4.
CMUK: what size dsp card would i need for 30 extensions and 7 SIP channels? also, by voice VPN i was thinking that there would be seperate VLANs for voice and data on a draytek at the remote site.
im not IT literate by any stretch of the imagination so maybe the infrastructure in my head isn't right.
OBTsystems: i'll need to look into the media relay, its not something ive ever used or been exposed to. The firewall is controlled by the converged Gamma router so i'd assume i'd need specific ports open?
also, have any of you used the Panasonic iOS softphone? again, its using a Gamma converged router so i'd need SIP username and password and SIP server ports (local and remote)?
Hello,
Firstly I'd get gamma to relax that firewall and give you a routed IP then get a Draytek 2862 and use that as your router into the cisco.
Every single time gamma have configured a cisco for me it's never been correct first or second time, additional to this I have had them randomly remove port forwards causing my customer to go down,
A DSP-S would sit well for your size.
In terms of your VLAN's, I mean it's up to you, I have lots of small customers with VPN's which pass voice with no VLANS.
Haven't used the IOS softphone unfortunately however if it's anything like other manufactures it will be a username and password with the public IP of the site unless you have a FQDN (unsure if it even supports that)
Port 15060 is for sip extension only. What phones are you using
If the system is on the latest firmware it shouldn’t let you save the media relay details with any port with 5060 in it. Ie 15060 25060 etc on the sip extension server. If you are using sip extensions on the media relay. You will need to do port translation from external port to internal on the system
Hi OBT, I have restarted the system but not the SIP router. i'll do both again.
I'm using KX-NT680 handsets.
The system is on 008.00055 which is the latest i think. It didn't like the 5060 or 35060 etc. I seen a similar screen shot on this forum so i put the 30000 in to be safe. I'm NOT using any SIP device.
Without sounding dumb, how do i do port translation from EXT to INT ports? I couldn't get a SIP device working on a mobile phone on a previous install - this is a first for me.
At our HQ, I have NS500 with a DSP-S card a static public IP Address.
Our branch have IP Phone model # KX-HDV130.
The phones shows registered but there's no audio on either side.
I have changed the following settings in the PBX Configuration :
Under Media relay :
NAT - External IP Address /FQDN - to my ISP Static IP
SIP Extension /UT Extension : NAT - SIP Proxy Server Port No : 49999 (Default was 0)
Under Port Number :
UDP port No.for Sip Extension Server : 49999 (Default was 5060)
I wouldn’t have a system on a public IP as you have no way of stopping someone hacking the system.
If you are leaving it on public IP I presume you wouldn’t use the media relay as it will write the headers as a nat.
I would strongly recommend putting a firewall on the system as the may not get a connection to the system but they could drown the system with requests and knock out the system
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