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3CX Remote Extension

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zconkle

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Sep 12, 2007
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Hello. I am using 3cx v3.1 for my office. I have a remote extension (3cx VOIP Client)I would like to use and have got it setup registered and can make calls on it. The problem I am facing is when I call out the other party cant hear me. I can hear them though. I know I can upgrade to 5.1 or 6.0Beta but these versions do not work well with my sip phones. Below is the server log when I call out from extension 104. I have got the stun server set up on the client end and the client machine is setup in the DMZ. I also have listed all the open ports for both ends.

Thanks in advance!

13:49:39.978 StratLink::eek:nHangUp [CM104001] Call(277): Ln:10000@CV1 hung up call; cause: BYE; from IP:209.120.140.17
13:49:31.337 CallLegImpl::eek:nConnected [CM103001] Call(277): Created audio channel for Ln:10000@CV1 (209.120.140.17:12008) with third party (76.185.76.17:3282)
13:49:31.337 StratInOut::eek:nConnected [CM104005] Call(277): Setup completed for call from Ext.104 to Ln:10000@CV1
13:49:31.337 CallLegImpl::eek:nConnected [CM103001] Call(277): Created audio channel for Ext.104 (76.185.76.17:3282) with third party (209.120.140.17:12008)
13:49:21.791 CallConf::eek:nProvisional [CM103003] Call(277): Ln:10000@CV1 is ringing
13:49:21.244 CallConf::eek:nIncoming [CM103002] Call(277): Incoming call from 104 (Ext.104) to sip:[email protected]:5060


Server Side Port List:


1 QRT-Server-RD TCP 192.168.1.40 Any 3389 3389
2 Ftp-Server-RD TCP 192.168.1.36 Any 4389 4389
3 NConkle-Accounting TCP 192.168.1.14 Any 5389 5389
4 Ben-RD TCP 192.168.1.13 Any 5395 5395
5 VOIP-Manage TCP/UDP 192.168.1.34 Any 5481 5481
6 Camera TCP/UDP 192.168.1.46 Any 80 80
7 QRT-NTServer TCP 192.168.1.22 Any 6389 6389
8 VOIP TCP/UDP 192.168.1.34 Any 9000 ~ 9499 9000
9 VOIPSTUN TCP/UDP 192.168.1.34 Any 3478 ~ 3480 3478
10 VOIP-RD TCP 192.168.1.34 Any 7389 7389
11 VOIP-RMT TCP/UDP 192.168.1.34 Any 5060 5060
12 VOIPTRP TCP/UDP 192.168.1.34 Any 42000 ~ 42019 42000 ~ 42019
13 VOIPTAPI TCP/UDP 192.168.1.34 Any 4300 4300
14 Audio TCP/UDP 192.168.1.34 Any 14000 ~ 14999 14000


Client Side Port List:

Port Range
Application Start End Protocol IP Address Enable
3478 to 3478 TCPUDPBoth 192.168.1.102
42000 to 42019 TCPUDPBoth 192.168.1. 102
5060 to 5060 TCPUDPBoth 192.168.1.102
4300 to 4300 TCPUDPBoth 192.168.1.102
14000 to 14999 TCPUDPBoth 192.168.1.102
443 to 443 TCPUDPBoth 192.168.1.112
to TCPUDPBoth 192.168.1.
to TCPUDPBoth 192.168.1.
to TCPUDPBoth 192.168.1.
to TCPUDPBoth 192.168.1.


Remote Extension Settings:

Other options <<
This section allows you to configure other options for this extension. More >
Extension is external. Enable the "Extension is External" option if this extension is located outside of the corporate network or resides on a different subnet.
PBX delivers Audio. You can configure a caller ID that should be set for calls from this extension. Note that many providers do not support this and this may be overridden or ignored. Toggle the "Supports Re-invite" option if call transfers are not working reliably for this extension. Some older SIP phones do not support changing call parameters during a call. Toggle the "Supports 'Replaces' header" option if call transfers are not working reliably for this extension. Some older SIP phones do not correctly implement this option.
ser Status allows you to define whether the user is available to take calls. If set to "Away", the extension will behave as configured in the "Phone Busy" tab of the "Destination Unreachable" section.

Extension is external *
Bind to Media Server
Supports Re-Invite *
Supports 'Replaces' header *
User Status AvailableAway
Queues Status Logged OutLogged In
Outbound caller ID 104
SIP ID ncampbell


Thanks Guys!
 
Last edited by a moderator:
you need to check this Bind to Media Server
 
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