SIP Calls get Dropped

Hi Everyone,

I’ve finally configured my Asterisk server how I want it, but I am
having a big problem which I can’t find a solution for. I tried posting this to Google Groups, and this is my first time on this forum.

I have one Polycom SoundPoint IP 301 phone and my Asterisk server on my
home network. One of my friends bought a Handytone 286 ATA for his
house. He can successfully register with my Asterisk server (using an
outbound proxy and STUN server). I am able to call him (x1137) just
fine.

The problem is that when he tries to call my Polycom phone (x1000), it
rings, but as soon as I pick it up, the call dropps. He also can’t dial
out to my ZAP trunk, or even to voicemail.

Any ideas what is going on? I know that there is some way to get it
wokring because I can call him, but I’m really stuck on this one.

Any suggestions would be great!

Thanks!

This may be an issue with your firewall, or it may be an issue with your sip.conf. It may be something totally unrelated to either, but I would suspect one of those. Can you elaborate?

My Asterisk server is on the DMZ on my router/firewall, so there should be no problems there.

Here is my sip.conf:
[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
allow=all
;disallow=all
;allow=ulaw
;allow=alaw
context = from-internal ;from-sip-external Send unknown SIP callers to this context
callerid = Unknown
dtmfmode=rfc2833

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

Here’s my sip_additional.conf:
[1000]
username=1000
type=friend
secret=3994
record_out=On-Demand
record_in=On-Demand
qualify=no
port=5060
nat=yes
mailbox=1000@default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=“My Name” <1000>
allow=all

[1137]
username=1137
type=friend
secret=1106
record_out=On-Demand
record_in=On-Demand
qualify=yes
port=5060
nat=yes
mailbox=1137@default
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=“Friend’s Name” <1137>
allow=all

And this is my sip_nat.conf
nat=yes
externip = 67.188.137.145
fromdomain = jamesonray.com
localnet=192.168.1.101/255.255.255.0

Thanks a million!