SIP to SIP

#1

I am very new to the field of asterisk. While configuring the xlite sip phones i m getting this problem…

When i configure two soft phones on the same computer both having different user names and auth names and i try to call between the two soft phones sometimes all the calls end up in a single phone while rest of the time calls that r made to one of the phone shows the following error :

“Unable to create a channel of type SIP” and all the users are busy.

If anyone can plz help me out i will be very thankful to u…i am unable to get out through this problem for last 6 days.

Thanks

#2

In general it is not a good idea to run multiple SIP endpoints on the same machine, or SIP endpoints in conjunction with an Asterisk server on the same PC. The issue is that you need to be careful about what you configure as the SIP and RTP ports and ensure there is no cross-over between the different clients. While it is possible to do this, you really need to watch what you are doing.

I would recommend using two different machines, test the X-Lite softphones, then if that solves the problem you know you have a config issue in terms of your SIP/RTP configuration on the same box.

#3

If you can’t do that, (because you just don’t have two computers available) you might just try to use a different SIP port for one of the instances of the softphone.

The SIP port that Asterisk will attempt to use to create a SIP channel is 5060. You can change that in many softphones and hardphones, as well as in the sip.conf file by simply putting port=5062 (for example) in the definition of the SIP peer that you would like to use a different port.

That should allow you to have multiple instances of a SIP softphone on your PC however, to successfully make use of them to send and receive audio, you may have to install a separate sound card.