Ipphones in remote location

[mbs1]

Hi,

I hope someone can help.

Xlite in a remote location ( behind a nat ) can register to the PBX at a remote location ( also behind a nat). The correct ports are opened on the firewall both ways. The remote soft phone can call a phone ( ipphone or xlite ) on the same subnet as the pbx, ringing is heard but on pickup the trace shows continious icmp messages. The trace also show the remote XLite as having resolved and sent its public IP address in the SIP Invite message. However the OK Contact message subsequently returned to the Xlite from the PBX shows the internal private ip address of the called party.The remote xlite then tries to send RTP packets to the private ip via the internet which fail and the ICMPs.

Has anyone got remote Extensions working. Would appreciate any help. Have got the RC1 ( 6 march ) version installed.

thanks

mbs

 

[mbs1]

hi travis,

thanks for sharing the msexternalinterface tip.

with this set to the public ip address of the firewall the pbx, i was able to have 2 way audio between 2 remote extensions and the remote extensions and my fwd viop gateway.what seemed to happen here was the pbx was not relaying the rtp via itself but telling each endpoint the others rtp ports. whilst this worked, the remote endpoints were not able to call the ipphone on same subnet and vice versa. there was ringing but no audio. the trace showed the pbx giving the remote endpoint a contact ip of the public ( what i would expect ) and the correct rtp port. however because the firewall was setup to port forward the range of rtp ports to the pbx, when the remote endpoint sent an rtp packet it got to the pbx but because the pbx does not proxy in this mode, it did not get to the ip phone and hence no audio( audio one way only). if the pbx was able to proxy for internal phones this would have worked. I verfied this by port forwarding the relevant rtp ports straight to the internal ipphone and calling the fwd. the echo test worked perfectly.

burt if we had 20 internal phones then we would have to set individual rtp ports for them and port forward each one individually which will probably be a big admin issue
i am hoping that the rc2 fix will allow the pbx to proxy calls correctly

hope this helps and i am right in what i understand

regards

mbs

 

[mbs1]

sorry travis

i missed out " as pbx " in line to "call the ipphone on same subnet and" ie remote extensions could not have 2 way audio with ipphones on the same subnet as the the pbx.

By remote, i mean 2 ipphones in a different location behind Nat Firewalls and different isp's.

thanks

mbs

 

[dekatech]

mbs,

we are on the same page....

i am able to use an external phone (behind seperate NAT on seperate lan) to register successfully to the 3cxserver and make calls to any outside line, and call into 3cxserver (leaving vmail message works, but ext to ext talking does not??), when trying to dial out from the 3cx server network to a remote ipphone, no ringing, but vmail eventually picks up....

wonder if mslocalinterface can be tricked?

 

[dekatech]

While sniffing packets I noticed I make a connectionion to the remote WAN from which the external ipphone gets its "nat" ed address. I get a "trying" message in my sip stack protocol, but then the remote IP is communicating back with a slimp3 protocol

Travis

 

[archie]

To clarify: RC2 has an option to bound extension to Media Server. This should be useful for external extensions, because our Media Server is able to communicate with clients behind NAT, as well as with local clients, thus performing proxy of RTP traffic between them.

 

[Anonymous]

It can work, as with all things it takes time. I am of the opinion if it has an IP you can use 3cx .

Perhaps this will help you, if you look at an SIP call this is how it goes.

Protocol
SIP Invite on port 5060
SIP Proxy Authentication Required 407 on port 5060
SIP Ack 2003 on port 5060
SIP Invite 2002 on port 5060
SIP Trying on port 5060
SIP OK 200 on port 5060
RTP RTP 4000 on port 9002
SIP Ack 2003 on port 5060
RTP RTP 4000 on port 9002
and destination port 10004
RTCP SenderReport 200 on port 10005
and destination port 9003
SIP SIP-Bye 2005 on port 5060

So you see that ports 900x are the ones you configured in 3cX for external calls (for all phones that have extneral ticked)
1000x are the RTP ports they look after the SOUND of your call eg your voice.

Most of the issues relate to NAT, and STUN will help to solve that. But I fix most with proxy, I have my 3xC box registered at each phone as a proxy aswell.

 

[mbs1]

Hi all,

just to let you know that with Rc2, ipphones in remote locations ( remote extensions) are able to ring and get 2 way audio with Central extensions and and voip gateway setup.

I have verified the following with a set config

a. 2 extension in locations on same subnet can dial and get 2 way audio
b. The same extensions can dial and get 2 way audio with fwd and voipcheap configured gateways.
c. They can also dial and get 2 way audio with extensions in remote locations which are also behind firewall and Nat.

d. An extension in a remote location can dial and get 2 way audio with a extension in the central site.
e. The same extension can also dial and get 2 way audio with fwd and voipcheap configured gaterways.
f. 2 remote extensions in a remote site can dial and get 2 way audio with each other.

I have looked at the traces and can see that the flow of Data and control is as per what archie and Costas say.

I am also able to leave a message and i can see a notification in my mailbox.

Thank You Archie, Costas,Itfarmer and Dekatech for your assistance

 

[archie]

Hi dekatech,

It looks like you have problems with redirecting incoming SIP messages to your PBX on router. You don't hear ringing tone on outbound call - that means SIP 180 Ringing response cannot reach you.
Can you sniff and see what 'Contact' address are passed in initial INVITE to external extension?
Anyway, try to bind external extensions to Media Server.

 

[vanDivX]

RTP RTP 4000 on port 9002
and destination port 10004
RTCP SenderReport 200 on port 10005
and destination port 9003


So you see that ports 900x are the ones you configured in 3cX for external calls (for all phones that have extneral ticked)
1000x are the RTP ports they look after the SOUND of your call eg your voice.

you confuse me, I understood from reading the manuals of the PBX and the SoftPhone that you just need 900x ports forwarded for voice - RTP (forwarded to 3CX PBX PC) and now you bring in 1000x ports, where does it come from?

the remote soft phone uses just 506x ports

3CX Phone also uses another UDP port speech transmission (RTP) with the value of additional 2 (i.e. if port 5060 is used for SIP, port 5062 will be used for RTP). Any further simultaneous calls use another port number, again increased by additional 2.

vanDivX