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- Sep 24, 2012
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Greetings,
I am testing a cloud based 3cx server setup. I have 2 grandstream phones. Each phone can place an out bound call. Each phone can receive an outside call. Each phone can call each other. I am using the SIP Proxy Manager. I have searched thru this forum to overcome a great deal of small aggravating issues and have done alot of tweaking and searching thru this forum just to get this far.
Now my new problem for which I cannot find an answer.
If phone 1 receives and outside call and I answer it there is audio both ways just like a normal call should be. However, when I transfer the call to phone 2, phone 2 rings and I answer it, but there is no audio on either end. The call does not end, there is just no audio. When I hang up phone 2, then the call ends. I assume this has to do with RTP ports, but I don't understand where the problem is? I would think the SPM should be able to handle this.
On a more general level, I am doing all of this thru the SIP Proxy Manager. Would it be better if I just used STUN and/or went thru the hassle of configuring the firewall to do port forwarding to each phone? I was experimenting with the SPM to see what it would do in hopes that it would simplify firewall changes/issues -- so I would appreciate any opinions on that from those that have successfully setup a remote server and a number of remote extensions.
Thanks,
Loren.
I am testing a cloud based 3cx server setup. I have 2 grandstream phones. Each phone can place an out bound call. Each phone can receive an outside call. Each phone can call each other. I am using the SIP Proxy Manager. I have searched thru this forum to overcome a great deal of small aggravating issues and have done alot of tweaking and searching thru this forum just to get this far.
Now my new problem for which I cannot find an answer.
If phone 1 receives and outside call and I answer it there is audio both ways just like a normal call should be. However, when I transfer the call to phone 2, phone 2 rings and I answer it, but there is no audio on either end. The call does not end, there is just no audio. When I hang up phone 2, then the call ends. I assume this has to do with RTP ports, but I don't understand where the problem is? I would think the SPM should be able to handle this.
On a more general level, I am doing all of this thru the SIP Proxy Manager. Would it be better if I just used STUN and/or went thru the hassle of configuring the firewall to do port forwarding to each phone? I was experimenting with the SPM to see what it would do in hopes that it would simplify firewall changes/issues -- so I would appreciate any opinions on that from those that have successfully setup a remote server and a number of remote extensions.
Thanks,
Loren.