Call termination problems and call routing query.

[Anonymous]

Hi,
I am able to create extensions, register them on the 3CX softphone, and make calls also. The problem comes when one of the parties terminates the call. Issues are mentioned below:
1) If one person ends the call, the other person's phone should also auto terminate the call, this does not happen.
2) The 3CX PBX server does not recognize the call termination and keeps sending UDP packets to both the parties(3CX softphone shows that no calls are active).

Apart from this, I read on the internet that SIP server is only used to initiate the call and then RTP communication is P2P. In 3CX implementation all the RTP or voice traffic is also going through the server (for internal to internal calls also). Is this intentional?

Do let me know if you need more information from my end to resolve this issue.

 

[N_G]

Calls

Hi,

Thank you for your feedback.

Are these calls going via a VOIP provider or a PSTN gateway? Or is it an internal call? What phone is on the other end?

 

[Anonymous]

thanks for your time... the issue is resolved now. the problem was with the 3CX softphone. when i was pressing the call termination button, the software was not sending the information to the server to action on. I found this via sniffing the packets. As per the info. available on 3CX site, i changed the client phone to X-Lite and things are better now. few issues still remain for example, i am unable to use the GSM codec for communication and all the call traffic is flowing through the 3CX Server(even for internal to internal calls).
I would prefer if the call traffic would go directly from caller to calee. Is there a way to achieve this?? I am sure most of your coorprate customers would prefer this as it would save some bandwidth!

 

[archie]

few issues still remain for example, i am unable to use the GSM codec for communication and all the call traffic is flowing through the 3CX Server(even for internal to internal calls).
I would prefer if the call traffic would go directly from caller to calee. Is there a way to achieve this?? I am sure most of your coorprate customers would prefer this as it would save some bandwidth!

Yes, GSM codec is already implemented and will appear in the next beta. About direct traffic, we're aware about it and this is intended behavoiur to test load capability of our Media Server. Also, it can help for the cases when caller and called parties have no common codec.
I think in release version we'll make direct RTP traffic for internal calls, or, at least, provide an option to turn this feature on/off.

 

[Anonymous]

Archie.... u are the best.. man! what u said was music to my ears.. just one last thing... when are u planing to release the final edition..? i know beta 3 is about to be relesed in a weeks time... wating to download and test it..