Can't Hear Sound From Hardphone ( Linksys SPA941 )

Asterisk Asterisk Support
1

Hello All,
I’ve been scouring the forums for two days looking for an answer to my problem, so I guess I’ll go ahead an ask for help ; )

I’m attempting to set up a test Asterisk system using AsteriskNow to replace my library’s current phone system. I’ve put together an Asterisk server with 4 FXO ports ( 1 is currently connected ), a Linksys SPA941 phone, and a laptop running Ekiga as a softphone.

The problem is that with the SPA941, I can speak into it and hear my voice from the other end, but it does not output any sound sent to it. This includes voicemail. With Ekiga, I can hear the voicemail just fine, but with the SPA941, it connects, the server says it’s playing sounds, but all I get is silence.

Any ideas? I’m wondering if this is a codec issue.

2

Type:
tcpdump udp portrange 10000-20000 > file

Into your console and press enter, make a broken call from your ip phone, then ctrl+c in your terminal.

Then open the file in a text editor and see where packets are coming/going to/from. Its likely the rtp stream (the voice data) on one leg of the call is going to the wrong place.

3

Type:
tcpdump udp portrange 10000-20000 > file

Into your console and press enter, make a broken call from your ip phone, then ctrl+c in your terminal.

Then open the file in a text editor and see where packets are coming/going to/from. Its likely the rtp stream (the voice data) on one leg of the call is going to the wrong place.

4

Thanks for the tip. I ran ‘tcpdump udp’ because the portrange gave me a syntax error.

Here is a copy of the output: http://kylehall.info/downloads/asteriskTcpdump

The asterisk server is 192.168.1.80, the phone is .81, and I was making an outgoing call to my cellphone. I could here my voice on the cellphone, but not on the sip hardphone.

Thanks for the help.

5

Have you tried testing it with asterisk’s echo test? This is my first port of call. The dump appears to show rtp data going both ways.

6

No, how would I go about doing that?

7

Add this into the context where your phones are in extensions.conf

exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Answer()
exten => 500,n,Playback(demo-echotest)
exten => 500,n,Echo()
exten => 500,n,Playback(demo-echodone)
exten => 500,n,Hangup()

Then dial 500

8

Dialing 500 and trying the echo test on the softphone works perfectly, but on the hardphone I don’t even hear the voice prompt even though the phone is connected, the call duration keeps counting up.

9

Post your sip.conf entries for both phones please.

10

I figured it out. I ordered an SPA942 and swapped it in for the 941, and it works perfectly! I must have received a bad phone. Anyway, thanks for all your help, I really appreciate the time you’ve spent trying to help me solve my problem.

Cheers.

11

No problem at all, I was beginning to think it was an odd problem! One way audio problems are usually evident after a packet dump and usually involve natted situations.