I am new to the Kirk 6000 product and I have hit a bit of a brick wall, and I was hoping to be pointed in the right direction if possible.
I have a very basic setup on test with Kirk 6000 server, 2 base stations and 2 handsets 5020 & 4040. I have literally unboxed the device yesterday and have performed some basic functionality tests. It soon became apparent that no Call Progress Tones were present on the 5020 handset. After upgrading to the most recent firmware I now have most of the CPT.
However I am still not hearing ring tone at the 5020 earpiece whilst making an outgoing call. Whilst I trawl through some SIP traces I thought I would post on here to see if anybody had any similar experiences.
Just thought I would put some more meat on the bones regarding this issue.
Having gone through the SIP traces it would appear that the Kirk Server and Extension are not sending RTP during the call setup. When the call is established between the Kirk and the far end, two way communication is achieved.
It would appear that when I presumed no ring back tone was being generated, I am actually experiencing one way audio.
Hopefully this will strike accord with somebody out there, and as mentioned earlier any help is appreciated.
The call progess tones are generated in the handset at a DECT level and they are generated based upon the SIP call progress.
In the case of SIP they will generate ringback when 180 Ringing is sent by the call handler. I have been told that the KWS6000 will also use the 183 early media however I've never seen this.
One thing you might have seen but maybe not is on the status menu in the KWS6000 you can grab some packet captures to see whats happening.
It is very kind of you to take time out to respond to me.
The 5020 handset that I was testing initially did not provide any CPT until I upgraded it to 08pa firmware.
I have used a combination of Wireshark and the Kirk's on packet capture to trace some failed calls. It would appear that our call handler is issuing 180's on calls between the Kirk's extensions. However when we break out to the PSTN network the call handler is issuing 183's instead of the 180's.
One of our engineers with more in depth knowledge of SIP than I, suspects that the Kirk is not sending RTP to our firewall during call setup. Then once the 183 responce comes back from the far end (PSTN) the firewall does not have an open port to connect the two parties to tie up ring back tone. Once the call is answered by the PSTN end RTP is fine between the 2 points, ie. two way audio.
Hopefully I have explained this clearly enough, and if you do have more time to think about it further I would be even more grateful.