Several troubles with Asterisk setting up

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.

  1. 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.

I can’t say why you may have no audio after calls are answered. Some reasons may include:

  1. Network problems
  2. A PC firewall program on the client PC
  3. An incorrectly configured client
  4. IPTables running on the Linux machine

Try shutting down any and all firewall programs. They complicate things.

As far as the audio quality goes, softphones are notorious for bad quality. Don’t expect the audio to be as good as a “hardphone”. You’ll only be disappointed.

Remember… the PAP2T is designed to be INTERNET facing. Not internal network facing. You’re asking it to direct it’s packets backward to it’s internal network, but it’s really supposed to be seeking an internet VOIP provider. For that reason, it may need to see the Asterisk by way of a NAT.

You can easily test that by simply taking the PAP2T off of your internet connection. Configure it so it’s internet connection is compatible with your own internal network, and connect the INTERNET connection to your local network switch. Then place some test calls.

You mention a hub… I hope you don’t mean an actual HUB. If so, throw it away and get a network SWITCH. You can’t do VOIP with a hub.

Well you can if you strive for packet loss and jitter. :smile:

Thanks for your help,
I figured out why I don’t have the audio. I used ethereal to analyze traffic on my network and I found out that SIP packets are flying correctly between calling parties (through asterisk) but RTP packets (the actual telephony data) are straying. They go from both clients to the PC with asterisk. I don’t think they are supposed toact like this. From what I know the RTP packets should be exchanged directly between the calling parties. Unfortunately I don’t have any idea how to fix this. Can you help me please?

That is not incorrect behavior, but it can be overcome.

You may want the Asterisk box to be in the middle between two endpoints to do transcoding. If one end point uses alaw, and another ulaw, Asterisk can convert the codec for you so the two endpoints can communicate.

If you want the packets to go directly between the two endpoints, what you want to do is set the endpoints, and the Asterisk, to support reinvites.

In your sip.conf file, add a line (in each sip definition) that says:


Check your SIP devices and make sure they are setup for reinvites as well.

In your sip.conf file, be sure you are allowing a codec that your endpoints can use, and that both endpoints agree about the codecs that will be used, or this won’t work.