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I had the same problem. It ended up
being some settings in sip.conf One of these settings did it for me (not
sure which one, as I added them all at once, then it worked): Port=5060 localnet=192.168.1.0/255.255.255.0
;<-----wutever the local subnet is (that the asterisk server is on) nat=yes externip=68.92.31.19
;<----wutever your public IP address is if you asterisk server is behind a
NAT firewall (don’t worry, the one listed is bogus; it’s not my
real addy) this is in my sip_additional.conf file
where 204 is the extension number of the remote extension: [204] username=204 type=friend secret=204 record_out=On-Demand record_in=On-Demand qualify=yes nat=yes insecure=very [EMAIL PROTECTED] host=dynamic fromuser=204 dtmfmode=rfc2833 context=from-internal canreinvite=no callerid=device <204> I’m guessing someone else will chime
in to say that these settings aren’t correct for everyone, but this is
just what worked for me. I have my asterisk server behind a linksys
wrt54g with the DMZ configured to go to the asterisk server. From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Brashear I am working with Xten lite for now. I am able to register
in but when I call out I can’t hear anything. The caller on the other end can
hear me just fine. Any ideas? I can get SIP to work fine internally. I also have all the ports open in the firewall including
10000 – 200000 -J |
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