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Transferring Calls via SIP

mikeS
Occasional Advisor

Transferring Calls via SIP

I have an environment of 30 NBX V3000s; all NBXs are at 6.5.22p05 and operating in SIP mode. I am able to do four digit dialing from main site to any remote site and the remote sites can four digit dial to the main site. We currently do not allow the remote sites to four digit dial another remote site.

 

My issue is when an outside caller calls a remote site and does not answer, we have that call transferred to our in house call center (main site) via the SIP trunk. This allows the outside caller to speak with a live representative. But when the main site transfers that same call back to the originating or any other remote site via SIP there is no audio in both directions and if you stay on the call long enough it will disconnect (hang up). Does anyone have any ideas?

3 REPLIES
merlin215
Valued Contributor

Re: Transferring Calls via SIP

You say SIP trunk , NBX SIP trunks ? Calling site to site ? 

 

So a call comes into site one , transfered to site 2, and then sent to site 3 with no audio ? 

 

Can you replicate this with normal calling by transferring outaside calls to different tels ? Sounds like something is either blocking the audio or the route to the tels via the router may be an issue after the third transwer attempt

 

Would prob need a trace of a good call and bad call as it sounds like something on the network possibly . When did this issue start occurring ?  All audio is RTP as in SIP mode we changed the Audio from NBX to SIP ( RTP ) . 

 

I don't think anything in set up could cause this . The network or asome sort of bug . Only a trace would show us as I have not heard this one at all . 

mikeS
Occasional Advisor

Re: Transferring Calls via SIP

When I referrence SIP trunk, I am referring to the trusted SIP end point that must be created. Also, the scenario actually reflects that the caller phoned in from the PSTN to the remote site; remote site does not answer the call is forwarded to the main site via SIP (site-to-site vpns are setup between the sites); the problem occurs when the main site attempts to transfer the call back to where the call originated. So, PSTN call transferred from remote site via SIP to main site , PSTN caller can hear main site. Main site attempts to transfer call back to same remote site via SIP. PSTN caller cannot hear originating remote site.
merlin215
Valued Contributor

Re: Transferring Calls via SIP

Just as I tohught ....

 

We would need a network trace and a case opened . I would need to see what is going on in the third leg of the call ( No audio after the final transfer ) .