Call Waiting

[Uday kumar]

Hi All,

I am using Freeswitch with sip.js. Everything working fine. Just I want to activate one facility for call waiting. For example if I am on call using Wephone and same anyone else calling to me then call should go on hold. If want to keep first call on hold and unhold second then I can do. This functionality working with JSSIP but not with SIPJS. 

Please help.

Thanks

Uday.  

[James Criscuolo]

Hi Uday,

  If you are talking about the ability to take multiple calls at once on a UA, this is definitely possible. Having a second INVITE arrive while you are already on a call should not automatically put anything on hold, but you can certainly do this yourself, albeit without renegotiation for the time being. I've never heard of a hard phone auto-holding, so having that as baked in behavior seems backwards.

James

[nit...@bitwisetechno.com]

Hi Guys

I am also getting the same issue , when the second call comes on the webphone it gives a 486 BUSY aand hangs up the caller .. Is there a way ..in which we can switch the calls in between two calls

Regards
Nitesh  

[James Criscuolo]

Hi Nitesh,

  Can you reproduce this with our demo-phone (note in only works in FF for the time being due to https)? It is possible that freeswitch is actually the culprit here, so it's important to check the console to confirm that SIP.js is indeed the originator of the 486 BUSY.

[nit...@bitwisetechno.com]

I tried using the demo-phone , its not getting registered on the server ..
same when I tried using other SIP webphone there also its taking more than one calls and allowing to switch..

Regards

[Uday kumar]

In my case B486 BUSY received from SIP.js. I have checked on browser's console.

[James Criscuolo]

For the demo-phone, you need to register it with your own server (wsServer). If you can provide step by step instructions with the demo phone, or a fiddle, of how to cause this, then I will attempt to reproduce and fix. Otherwise, we have a production application that behaves exactly as you expect, and a 486 only gets sent in one seemingly rare case according to the source, so I will not be investigating this myself for the time being.

, 

[nit...@bitwisetechno.com]

Hi

Thanks for the quick reply

For recreating the issue we used the demo phone with our sip server ( freeswitch with WSS ) but the request is not coming on the server logs .. not able to register the phone .

Regards

[James Criscuolo]

There is only one place in our source where we send a 486, and this isn't a situation in which that line can be hit, so I don't think it's anything directly with the library. Are you both working from some example somewhere, or is there somewhere in your app that a 486 gets sent?

Another case where something quick and hard to catch occurs is if getUserMedia is automatically rejected, for one reason or another. Putting extra logging around your getUserMedia call may uncover something.

James

[Nitesh Sharma]

Hi

Thanks for the reply .. yes I checked it was not from the library , we had to call it . so we returned 180 and now its returning ringing event,

Thanks again !!

Regards