RC2 and softphone connection from external source

[spamoni]

I recently upgraded to RC2 of 3CX and am having this issue:

There is no issue when I connect a softphone from an site external to my server. However, when I dial an extension on the PBX, I get no audio or ringing. The extension I am calling is not connected as I wanted to hear the voicemail prompts - but nothing happens.

Can someone please let me know what I have to set on my Router or somewhere to allow me to hear the audio.

BTW: if I do this behind the firewall, everything is fine - there is no issues with audio.

my log shows the following:

06:35:28.343 ServRegs:nRemove Unregistered: Ext.1375
06:35:25.406 StratInOut:nHangUp Call from Ext.1375 to 999 has been terminated
06:35:15.437 CallLegImpl:nConnected Established media channel for Ext.1375: remote=10.30.50.1:40516; local=192.168.100.5:7002
06:35:15.421 CallConf:nIncoming Incoming call from Ext.1375 to sip:[email protected]
06:35:06.625 StratInOut:nHangUp Call from Ext.1375 to 999 has been terminated
06:34:50.421 CallLegImpl:nConnected Established media channel for Ext.1375: remote=10.30.50.1:36316; local=192.168.100.5:7000
06:34:50.000 CallConf:nIncoming Incoming call from Ext.1375 to sip:[email protected]
06:34:33.796 ServRegs:nAdd Registered: Ext.1375

 

[Kevin Attard Compagno]

It appears that your external softphone is reporting a private IP address (10.30.50.1) - did you set up STUN resolution for the external softphone?

Also ensure that for the external extension, advanced options "Device is External" and "Bind to Media Server" are both enabled.

If this still does not resolve your audio issues, you may have to configure some sort of port forwarding on the remote extension's firewall.

 

[spamoni]

The IP is from my work - which could be an issue. I am going to try it another way later tonight and it does work internally, as long as everyone is local to my network.

BTW: This PBX is great, even with these questions. It is much easier to stup than asterisk, and being new, I would expect some issues as asterisk did as well. Keep up the good work.

Now...

I did the following:

1. added stun.3cx.com as the stun server
2. checked the recommended boxes for the extension (not sure if requires a restart of the pbx)

Still not working.

Question:
what ports do I need open on my router and redirect to the PBX?

5060 for SIP?
Do I need to set 9000+ for external extensions?
Do I need to forward 10000-20000 for sound?

Also - if I setit up as an external extension, will everything still work internally - extension to extension dialing? I think it should, since I think this designates the port to use, correct or not?

Thanks

 

[3CXsupport]

Hi,

Thanks for the words of encouragement !

If you use a phone as external you need to open the 'external range' on your firewall to forward to the PBX, by default this is 9000-9003. You mught need to widen that range if you use several phones. You also need to publish 5060 to the PBX.

On the other hand your external phone is going to send an INVITE to start a conversation to the PBX and it is going to tell it which ports it (the pone) is listening on. The PBX will start sending RTP (audio) to those mentioned ports. These have to be 'published' on the router (that one that the phone is behind).

i.e. if your phone is set to use RTP range (e.g.) 32000 - 32100 your router will have to know what to do when if receives a packet on a port in that range. Therefore when the PBX starts sending audio the router will forward it correctly. This actually depends on your type of router, but if you are having issues it's best to configure in the described manner.

btw oni as in spamdemon ?

 

[spamoni]

actually - name given was a nickname given to me by a family member - being italian and all. spamoni should really be spumoni (the italian ice scream). LOL

I guess I should have rethought the sname huh...

 

[spamoni]

Ok...

I did some more diging. I went into my VOIP setting and disabled the following:

Device is external
Bind to Media Server

I also made sure they were not enabled on my extensions. Now I can hear myself from a landline to the softphone and vice versa.

However, I need to get the Music on hold working - I suspect I need to atleast check the Bind to Media Server.

So in my case - nothing changed from my previous info, I am still testing from work and it was finally successful. The only change I made wa to disable the 2 items listed above.

 

[spamoni]

Still an issue...

I tested tonight as indicated. The phone actually worked flawlessly. I connected it to my Viatalk provider and made a call. The quality of the call was great.

But the music on hold did not work. Is there a way to allow the music on hold to go through the normal ports - the ones you speak on?

I ask because it is my belief that if I can talk and hear someone we should be able to get the music on hold to them as well. I tested as follows:

1. setup 2 extension, mine and one for my buddy who lives about 30 miles from me.
2. connected via softphones connected to 3CX. was able to talk but was not able to hear the music.
3. used my softphone to call his landline through his PBX. so I was using 3CX and his landline connects to his asterisk box.
4. Music on Hold did not work, however, his music on hold did?

I have 9000-9007 forwarded to the 3CX as well as 5481 for management and 5060 for softphone connectivity.

what else can I try?

Thanks in advance

 

[archie]

Hi,

To provide music-on-hold your client should be connected to our Media Server. We can not play music pretending to be your phone! So, we need to redirect your client to MS. To do that we utilize so called re-INVITE, which provides client with new parameters for media stream. If the client doesn't support re-INVITEs, or does it incorrectly - we can not provide music no matter how we try. So, either your client should be bound to Media Server (no re-INVITE is necessary in that case), or it should support re-INVITE -- these are necessary conditions for music-on-hold.

 

[spamoni]

ok, so I will turn off the re-invite and turn on MS binding.

when I had MS binding and ReInvite on, they could not hear me, but I could hear them.

Anything I should be concerned about

 

[archie]

If you will have audio problems - please, provide us with piece of logs as you did in the first post on this thread.