Posts made by IT-ADMIN

i have windows clients (xp and 8 all have issues), i wil do what you tell me when i get home, because now i'm in the office ( behind the openvpn server)
see you later

pfSense is a very powerful firewall, it provide many services in form of packages, including proxy server, openvpn, dhcp, traffic statistics .....and more and mor, all in one, it is really a very good firewall

hi guys,

i have pfSense installed as firewall and openvpn Server, just before everything was ok and vpn clients get connected without any issues, just recently i encountered a problem: the client get connected and get ip (10.10.10.6), everything seem ok, but the client cannot neither ping remote ip address nor the ip address of the openvpn server (10.10.10.1) , when check the client openvpn virtual interface, i remarked that he don't have any gateway (which is supposed to be 10.10.10.1), but this problem happen just recently, before everything was ok

any suggestion ???

@scottalanmiller greaaaaaaaaaaaaaaaaaaaaaaaat, that's what i'm looking for, a bunch of experts willing to help and share their knowledge with other people,
really people will appreciate your work, thanks

thank you very much, see you soon Mr Scott (our VOIP expert) with a new topic

thank you guys for your input

@scottalanmiller

Yeah Sir you are right, i think that my firewall pfSense can do that because by way of it i can make queues, traffic priority, QOS, by way of these tools i think i can ensure that real time traffic have the highest priority than other type of traffic,

see you Mr Scott,,,,,,thank you

@scottalanmiller

but i have no control over the wan traffic, after the packets come out from the wan interface i can't control it !!!

great, is there any other thing that affect QOS ?? separating voice and data (VLANs) maybe a good idea (hhhhhh i know that you hate VLANs) i'm joking Mr scott

have a great day Sir

aaah you mean i select only one codec (g.723)
ok i got it
thanks

unfortunatlly i don't have this option of preference,
which are the other codec that i have to select with this codec, can i select it only or other additional codec has to be selected alongside ???

thanks

i tried to select only GSM(3) since it is compatible with low bandwith but the call cannot be established, so i selected the additional 2 codec

hiiiiiiiiiiiiiiiii Mr Scott, how are you doing

first of all, the available audio codec that i have in my PBX system is:
PCMA (8)
PCMU (0)
telephone-event (101)
iLBC (98)
G722 (9)
GSM (3)
SPEEX (97)
SPEEX (100)
SPEEX (108)
L16 (103)
L16 (11)
L16 (10)
G728 (15)
G723 (4)
G726-16 (104)
G726-24 (105)
G726-32 (106)
G726-40 (107)

i select only 3 : PCMA (8) + PCMU (0) + GSM(3)

best regard

Hi guys

i was unclear in my previous posts, what i have to say is :
when a sip client 192.168.1.10 want to talk with sip client 10.10.10.6 the following happen :
since we are connected via openvpn tunnel no NAT happen actually, because i run wireshark in both machine (remote vpnclient and IP PBX) and i saw that the packets carry the real IP of the sip phone not the public ip (source ip : 10.10.10.6 --- destination ip : 192.168.1.10)
what happened actually is routing not Nating, because there is a virtual tunnel between vpn client and vpn server and the vpn server route the packets to it destination (in this case 192.168.1.10 )
so the RTP traffic should be established without any issue since there is not NAT in the tunnel

i find the solution and now i have 2 site connected via openvpn and both location can call each other, the problem occur because i had wrong setting (in Ozeki server), before i set Discover public IP address using STUN server: stun.ekiga.net, this what created problem for me, i change it to my local IP PBX (Ozeki server) then everything work fine except sound quality, both side can hear each other but the sound was not totally clear (mixed with some noise),

any suggestion to improve sound quality knowing that in the main office the download speed is 4 mbs and upload speed is 0.5 mbs , and in the branch office we have download speed is 10 mbs and upload speed is 2 mbs ,
i think that my connection speed which decrease sound quality ????? or maybe something else ??? any suggestion ??

thank you all for you answers

i'm very appreciated really
thanks alot for your time
i will try my best to get that done

see you again soon

best regard

so what i understand is: i should configure pfs to forward VOIP traffic to it destination without any NATing,

No Mr Scott
unfortunately if 192.168.2.x want to talk to 192.168.1.x the following will happen : 192.168.2.x hit pfsense then pfsense route into 10.10.10.x then 10.10.10.x hit 10.10.10.1 (gateway of the branch pfsense) then 10.10.10.1 forward the packet internally into 192.168.1.x
soooooorry if i was not clear

in the beginning the branch box address the main box with it public ip, once the main box validate the key installed in the branch box the tunnel is forwarded automatically to openvpn interface which is virtual, in this virtual interface all port are opened
i hope i was clear

i have 2 different networks 192.168.1.0 in the main and 192.168.2.0 in the branch, so each pfsense box route between 2 different network

i'm using OpenVpn so i have a dedicated virtual NIC for VPN and all port are opened in this virtual NIC, as for RTP, how can i routed it correctly because really i didn't make anything regarding this protocol

best regard