external extension audio problem

[fala70]

I've an audio problem when I call an external extension. The audio work good only a direction, from internal extension lan to external extension. From external to internal not work.

The port forworded side server are all correct (192.168.1.100 is the server):

1 sip 5060 5061 192.168.1.100
2 RTP 8000 8000 192.168.1.100
3 HTTP_Server 80 80 192.168.0.100
4 esterne 9000 9500 192.168.1.100
5 3CX WEB SERVER 5480 5482 192.168.1.100
6 3CX 3478 3478 192.168.1.100
7 - 5072 5072 192.168.1.100
8 - 8888 8888 192.168.0.100
9 - 3456 3456 192.168.0.100
10 - 4662 4662 192.168.1.105
11 - 4672 4672 192.168.1.105
12 - 4711 4711 192.168.1.101
13 3CX 7000 7499 192.168.1.100
14 3cx 10000 20000 192.168.1.100

side external extension I use a hardware phone directly connected to a pc with a modem, without router and firewall.

this is the log of a call, 100 is then external extension, 101 is the normal extension.


18:40:52.406 StratLink:nHangUp [CM104001] Call(47): Ext.100 hung up call; cause: BYE; from IP:217.202.74.235
18:40:27.937 CallLegImpl:nConnected [CM103001] Call(47): Created audio channel for Ext.100 (217.202.74.235:62882) with third party (192.168.1.102:9000)
18:40:27.937 StratInOut:nConnected [CM104005] Call(47): Setup completed for call from Ext.101 to Ext.100
18:40:27.937 CallLegImpl:nConnected [CM103001] Call(47): Created audio channel for Ext.101 (192.168.1.102:9000) with third party (217.202.74.235:62882)
18:40:24.578 CallConf:nProvisional [CM103003] Call(47): Ext.100 is ringing
18:40:21.984 CallConf:nIncoming [CM103002] Call(47): Incoming call from 101 (Ext.101) to sip:[email protected]:5060



Any idea. Thanks to all.

 

[5qg4]

Please ticked following option for all external extension(s) under Other options at extension setting:

Extension is external
Bind to Media Server

For extensions at same LAN or VPN just ticked "Bind to Media Server"

 

[fala70]

Now I've activate all two options for both extension. There is again the problem. Now no audio for all direction

07:46:57.640 MediaServerReporting::RTPReceiver [MS105000] Call(4) Ext.100: No RTP packets were received on 00000008@:remoteAddr=217.202.104.128:54200,extAddr=87.6.59.135:10233,localAddr=0.0.0.0:9010
07:46:53.578 StratLink:nHangUp [CM104001] Call(4): Ext.100 hung up call; cause: BYE; reason: SIP;description="User Hung Up"
07:46:34.140 ServRegs:nAdd [CM113002] Registered: Ext.101
07:46:32.406 ServRegs::checkExpiration [CM113000] Registration for sip:[email protected]:5060 has expired
07:46:25.531 CallLegImpl:nConnected [CM103001] Call(4): Created audio channel for Ext.100 (217.202.104.128:54200) with Media Server (87.6.59.135:10233)
07:46:25.531 StratInOut:nConnected [CM104005] Call(4): Setup completed for call from Ext.101 to Ext.100
07:46:25.531 CallLegImpl:nConnected [CM103001] Call(4): Created audio channel for Ext.101 (192.168.1.102:9000) with Media Server (87.6.59.135:10231)
07:46:21.546 CallConf:nProvisional [CM103003] Call(4): Ext.100 is ringing
07:46:16.968 CallConf:nIncoming [CM103002] Call(4): Incoming call from 101 (Ext.101) to sip:[email protected]:5060

 

[fala70]

I tried also to ticked extension is external only for one, the audio work on for a direction, and this is the log:

07:59:02.921 MediaServerReporting::RTPReceiver [MS105000] Call(6) Ext.100: No RTP packets were received on 0000000C@:remoteAddr=217.202.104.128:48488,extAddr=87.6.59.135:10299,localAddr=0.0.0.0:9016
07:59:02.906 StratLink:nHangUp [CM104001] Call(6): Ext.101 hung up call; cause: BYE; from IP:192.168.1.102
07:58:51.062 CallLegImpl:nConnected [CM103001] Call(6): Created audio channel for Ext.100 (217.202.104.128:48488) with Media Server (87.6.59.135:10299)
07:58:51.062 StratInOut:nConnected [CM104005] Call(6): Setup completed for call from Ext.101 to Ext.100
07:58:51.062 CallLegImpl:nConnected [CM103001] Call(6): Created audio channel for Ext.101 (192.168.1.102:9000) with Media Server (192.168.1.100:7004)
07:58:49.640 CallConf:nProvisional [CM103003] Call(6): Ext.100 is ringing
07:58:45.734 CallConf:nIncoming [CM103002] Call(6): Incoming call from 101 (Ext.101) to sip:[email protected]:5060

 

[Mirzab]

Internal to external, audio OK. External to internal, no audio. Must be ports.

List of default ports used by the 3cx phonesystem:
5060 (stanard sip port) TCP/UDP
5061 (secondary sip port) TCP/UDP
5480 (portgre) (only on local machine)
5481 (apache) (only on local machine)
5482 (media server) (only on local machine)
3478 (STUN Server) TCP/UDP
7000-7500 (Default port for internall calls) (only on local machine)
9000 - 9003 (ports for external calls through VOIP Gateway) UDP
10000-20000 (RTP ports) (Used if no stun available for audio)

http://www.3cx.com/forums/the-portforwarding-faq-hope-this-gets-a-sticky-1697.html
Are you ensuring TCP / UDP as described? And for RTP which are you using?

 

[fala70]

Internal to external, audio OK. External to internal, no audio. Must be ports.

List of default ports used by the 3cx phonesystem:
5060 (stanard sip port) TCP/UDP
5061 (secondary sip port) TCP/UDP
5480 (portgre) (only on local machine)
5481 (apache) (only on local machine)
5482 (media server) (only on local machine)
3478 (STUN Server) TCP/UDP
7000-7500 (Default port for internall calls) (only on local machine)
9000 - 9003 (ports for external calls through VOIP Gateway) UDP
10000-20000 (RTP ports) (Used if no stun available for audio)

http://www.3cx.com/forums/the-portforwarding-faq-hope-this-gets-a-sticky-1697.html
Are you ensuring TCP / UDP as described? And for RTP which are you using?

Yes I am sure. I am using 9000-9005