3CX Remote Stun IP Phone all config

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KhaledQuenawy

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please could any one tell me what is all config i should do if i want to provision remote ip phone with stun method ...
1- port forwarding on the router where 3CX Server connect to
2- port forwarding on the router where remote ip phone connect to
... every thing ...
(the firewall checker is done .. and i make port forward on the remote router to the ip of the ip phone for ( 5065 ) TCP/UDP and (14000 - 14019 ) UDP
should i make this port forwarding too on the router that 3CX server connect too or not and sip alg is disable on both router ..
but still no audio on this remote ip phone
 
For multiple STUN endpoints we would need:
  • Each endpoint on its own statically assigned IP (whether it be static or via DHCP >> MAC assignment).
  • SIP ALG, SIP Helper or Deep Packet Inspection needs to be disabled on the firewall – can be different names based on the vendor (and can be GUI or CLI command).
  • Map ports correctly (port forwarding) to avoid sporadic ports for SIP and RTP being allocated
Example starting at port 5062 then increment to 5064, 5066 etc.

Each phone needs to have a unique - specific SIP and RTP/media ports – you can pull the extensions CSV file from 3CX and use it as a guideline to see what the advised STUN ports will be (when extensions are configured on the PBX).

SIP should incremented by x2 ports and RTP by X12 for Yealink phones normally.

I still do not recommend or support this setup and would always recommend a VPN or SBC.
 
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Example attached here.
 

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Do you have static ip on remote phones ?
Do you have several phones in STUN on remote location ?
For now , is the phone using 5065 sip port able to be provisionned properly?

see on this previous answer for STUN
https://www.3cx.com/community/threads/stun.65555/#post-280289
what you mean by static for ip phone .. you mean static ip for each phone in private range like 10.0.0.0 or 192.168.0.0 yes each phone have one because i make port forward for port 5065 and 14000 - 14019 and increase by 20 port for each phone
 
Each phone must have an IP address which does not change - so either static, or DHCP hard linked to MAC address.

SIP increment by 2 and Media/RTP by 12.
 
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