Good day to you all,
I am setting up my own small LAN where PC1 has Asterisk installed on it, PC2 has SJPhone telephony client and using Linksys PAP2T I have connected analog phone to my network. All of these devices are connected to a hub. At this stage I only want my two clients to be able to call with each other. I came accross several problems and I was wonderin whether somebody would be able to help me.
- When I am calling, the client is ringin but after i pick up the phone I cannot hear the voice. Echo test works fine on both sides, I do not have NAT, RTP port range is set in rtp.conf to 10000-20000. What should I do please?
2)From the analog phone I am able to initiate the call (ie ring PC2), but when I try to call from PC2 to the analog phone I get message from Asterisk that the extension is “circuit-busy” I tried to connect PC3 instead the analog phone and I was able to make the call from PC2 to PC3 and vice versa (in fact i get only the signalization that the extension is ringing…see my problem no.1) Should I get a firmware for the Linksys PAP2T?
3)When I try to use some of the sample sound files which Asterisk is provided with (they are in gsm format) I got very horrible quality on my PC, on the phone it is fine. Is there any way to improve this? Or is it because the gsm codec compression?
Thank you everybody for any ideas you may think out. It would help me a lot.