Intial problem


#1

I recently became interested in Asterisk@Home & for one reason or another I can’t get anything to work. I bought the X100P FXO card, (2) Grandstream phones, & loaded Asterisk 2.8. I’m able to login into AMP but according to this book I d/l I should have a Maintenace tab up top next to Setup. I should also have a Incoming Calls configuration selection on the left. What I’m thinking is maybe the diagrams in the book are from a older version os Asterisk@Home which is cool if thats the case. How can I add those options if they are needed for 2.8?

Here is my biggest problem now though, I can’t communicate to the other phone. I get a 503 error anytime I attempt to make a call, not to mention once I make my IP configuration changes on the phones, I lose the artifical dial tone.

I’ve been reading web resources & forums like crazy with no avail. Any suggestion or any advice anyone can give me?

Thanks in advance.


#2

Hi,

I haven’t ran 2.8 yet but I’ve been reading the release notes and the team at AAH have replaced the maintaince page with freepbx.org plugins. It’d be worth stopping over there and seeing how freepbx does the things you need it to do.

For the phone configs i would recommend dbl-checking your sip.conf configs compared to what you have on the phones (usernames, secret, ip allocation etc).

Worth having a read of asteriskguru.com/tutorials/g … phone.html if you haven’t already.

If there’s still no joy post your sip.conf and we’ll see if there’s anything there that needs tweaking.

Cheers

Stephen


#3

I found that link very informative. I was able to get dial-tone back now on both phones, but I can’t dial internally. Extensions 100 to 101.

Below is the generic basic sip.conf I’m using now. I had added some other commands in it before, but it seems like once I deleted them I got dial tone back.


; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.

[general]

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

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


#4

any suggestions?


#5

I suggest you go ask on the AAH forum, you’ll get a lot more people who use AAH to help you out given you have specific AAH related questions.

p


#6

I thought this was the AAH forum! If not, where is it?


#7

Speaking from my experience, the CODEC mismatching maybe is the cause of 503 error. it is usually related to the audio codec not being supported by one end.
if your phone support GSM, you can add it into your sip.conf,
allow=gsm
or you can use a softphone for the test, which usually support more codecs and has detail sip log messages.