Remote phones don't transfer sound

Asterisk Asterisk Support
1

Hello everyone. I have recently installed an Asterisk 1.4 server in one of our offices. Everything seems to be going smoothly except we have 2 remote sip phones connected to this server that don’t send or recieve sound. One is a Polycom Soundpoint 500 and the other is an X-lite softphone. Both connect to the server fine, and you can dial extensions and outside lines, but when the other end picks up there is no sound being transmitted. This is also the case when checking voicemail. I have the port range 10000-20000 forwarded from the router to the server. I have set that range in rtp.conf. I never mess with iptables. But I’m still not having any luck. It almost seems like even though I set the range in rtp.conf sometimes rtp packets will go through other ports. At least thats what I gather when I do a rtp debug, but I’m not sure. Here are some of my configs. Let me know if you need more. Thanks in advance.

sip.conf

[general]
port = 5059-5061
bindaddr=0.0.0.0
context=default
tos=none
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=alaw
allow=ulaw

externip=x.x.x.x

[test]
type=friend
callerid=test
secret=test
host=dynamic
disallow=all
allow=ulaw
canreinvite=yes
nat=yes
context=internal
dtmfmode=rfc2833
qualify=2000
pickupgroup=1
callgroup=1

rtp.conf

[general]
rtpstart=10000
rtpend=20000

2

Check the configuration of your polycom phone and softphone for an RTP setting and make sure it is using the same range your server is using. For instance, Grandsteam phones have a local RTP port entry that defaults to 5004.

If you can’t get your SIP based phones to work, I would recommend using IAX based phones for remote users which are better at traversing NATs.

Hope this helps,
Grizzlyism

3

Well I did as you said and changed the rtp range on the devices themselves and still no sound. If anyone has any other options I will be more than willing to try them out. If there are no other ideas, could someone please give me a recap of all the ports that are needed for a remote sip phone to connect to an asterisk server. At least then I can check the router and make sure everything is good on that end. Thanks again.

4

which phone is for test friend in sip.conf, Polycom or X-lite?

5

The test account is for the x-lite