No Voice on remote IP Phone

[digitel]

Hi All
New to this forum , I've searched along posts with no results....

I have latest 3cx ver. installed
All works ok

I set up an IP Phone ( Grandsteram BudgeTone 100 ) to be used on any remote site capable of an internet connection and a DHCP-Server router.

At first test, the ip phone registers correctly to 3cx server, it can place and receive calls (telephones ring) but no voice is present at hook-off, both directions

On 3cx router side, port 5060 is opened, both tcp and udp and forwarded to 3cx LAN IP
3cx server is reachable by a dyndns domain not a fixed public IP
When the phone rings and ,at hook-off no voice is present, I see however a continuos traffic on routers wan status led


Any suggestion or solution about issue ??

Thank you very much

Federico

 

[sigma1]

Most of the time that is due to a firewall issue if you followed all the setup directions.

What router do you use at the remote site?
Please make sure you specify the STUN server as IPORT
Enable Symmetric RTP

This, provided that you tested with the 3CX softphone at remote site and all is OK.

At the remote site, please do NOT FWD any ports, simply let the ports be.

 

[digitel]

What router do you use at the remote site?
Please make sure you specify the STUN server as IPORT
Enable Symmetric RTP

.

Sigma1, thanks for reply

I don't remember the exact router model, it is a USR wifi adsl one,
Where sould I specify the STUN server ? On Ip Phone setup ??
If yes, what stun server shall I choose , as I don't use any provider but only 3cx for communications between its extensions ???

I will look at Symmetric RTP and double-check with 3cx softphone.

At the remote site, please do NOT FWD any ports, simply let the ports be.

.

remote site router hasn't been modified,
Local 3cx cisco router has port 5060 forwarded to 3cx server

let me know

Thank you again

Federico

 

[digitel]

A further info.....
This is 3cx call log
301 is the local (LAN) calling extension
305 is the remote extension (grandstream ip phone) , registered to 3cx via internet.....


14:40:53.119 [CM503020]: Normal call termination. Reason: Server Failure
14:40:53.119 [CM503016]: Call(12): Attempt to reach "GrandStream"<sip:[email protected]> failed. Reason: Server Failure
14:40:53.119 [CM503003]: Call(12): Call to sip:[email protected] has failed; Cause: 503 Service Unavailable; warning: ; internal
14:40:53.019 [CM503025]: Call(12): Calling Ext:Ext.305@[Dev:sip:[email protected]:5060]
14:40:52.688 [CM503004]: Call(12): Route 1: Ext:Ext.305@[Dev:sip:[email protected]:5060]
14:40:52.678 [CM503010]: Making route(s) to "GrandStream"<sip:[email protected]>
14:40:52.678 [CM505001]: Ext.301: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA942-6.1.3(a)] PBX contact: [sip:[email protected]:5060]
14:40:52.658 [CM503001]: Call(12): Incoming call from Ext.301 to "GrandStream"<sip:[email protected]>
14:36:11.754 [CM503016]: Call(11): Attempt to reach "GrandStream"<sip:[email protected]> failed. Reason: Server Failure
14:36:11.704 [CM503025]: Call(11): Calling Ext:Ext.305@[Dev:sip:[email protected]:5060]
14:36:11.384 [CM503010]: Making route(s) to "GrandStream"<sip:[email protected]>
14:36:11.354 [CM503001]: Call(11): Incoming call from Ext.301 to "GrandStream"<sip:[email protected]>
14:29:59.759 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 96.9.132.83:3478 over Transport 192.168.192.3:5060
14:25:19.266 [CM503016]: Call(10): Attempt to reach "GrandStream"<sip:[email protected]> failed. Reason: Server Failure
14:25:19.166 [CM503025]: Call(10): Calling Ext:Ext.305@[Dev:sip:[email protected]:5060]
14:25:18.856 [CM503010]: Making route(s) to "GrandStream"<sip:[email protected]>

 

[sigma1]

I have a funny question...

What is the IP address of your remote router (WAN) and what is the IP address of your remote phone?

Are you running this over VPN?