Failure with clean installation - receive 603 declined from PBX

FreePBX Installation / Upgrade
21

I have explained to you what you need to do to get help. You don’t want to participate in the process.

I will repeat, only 2 probable and three possible reasons for the 603:

1 - Incorrect network configuration (NAT)

2 - Mismatch of CODEC’s in SDP (you would see this if you turn SIP debug off and turn regular debug on ‘sip set debug off’ then ‘core set verbose 0’ and ‘core set debug 128’

3 - The router on the remote end is mangling packets.

I don’t know what else you expect from us.

22

first of all, you were not that clear in your previous suggestions.
this is the first time that you give me exact command to turn on debugs that are required.

I will try these and try to see what’s wrong in SDP, codecs etc.

about “incorrect network conf. /NAT” :
I’m sure this is not the case. I told you the reasons. there is no nat.
but still I don’t know what is a "good network configuration"
is there a clear example of this ?

about “router on remote end mangling packets” :
I have a working Asterisk+FreePBX configuration , using the same remote end. The new setup is just newer Asterisk on new OS.
So the problem can not be about remote end.

23

Let me ask you this, and I am curious because you feel left out. Do you think these debug commands are hidden? The aren’t even FreePBX commands, it’s just Asterisk, You should be familiar with www.asteriskdocs.org every command is documented in detail. You will also see the sample sip.conf. The settings in FreePBX are Asterisk settings so you should use this as a refernce for sip settings module.

24

if I do ‘sip set debug off’ then ‘core set verbose 0’ and ‘core set debug 128’, I get no debug output on the console.

weird ?

25

I uninstalled everything.
Installed Asterisk from scratch and then FreePBX 2.8 on top of it.
All configuration and databases were removed prior to installation.

After the installation I jyst created 1 extension, 1 SIP trunk and 1 outbound route.
That’s all.

It is very simple. But every outbound call fails. (again the same way)
There is no outbound packet to the external SIP trunk.
Yet my extension receives a 603 DECLINED message.

This situation is really very weird.
isn’t this a simple setup ? Why doesn’t it send out any INVITE message ?

26

It won’t send out an invite if it declines the invitation.

You have something basic and simple wrong.

27

actually as seen in the logs, it checks for the outbound trunk, finds it , then declines.
Like you said, it is something very simple but I can’t find it.
How can I identify what’s wrong ?

28

ok, I tried a dirty path.
installed the same Asterisk version as in my old installation.
Then copied all configuration files for Asterisk and FreePBX to the new system.
Copied also the mysql DBs.

Now it works. Every call flow works with just 2 problems (maybe related):

  1. I can’t see the new calls on CDR, recordings or call logs of FreePBX.
  2. On GUI I see this error:
Retrieve conf failed to copy file(s) from a module's agi-bin dir: chmod(): Operation not permitted

I tried to change file permissions in various paths but couldn’t succeed.
How can I solve these 2 issues ?

29

couldn’t solve this new issue.
it must be something easy, can anyone comment ?

30

If you read the entire post, you’ll see that I tried to install newer Asterisk and FreePBX.
But that setup does not enable me to make ANY calls.

So I installed the version which works on my older setup.
But now the problem is not a big one. It’s just about correct file permissions.

Retrieve conf failed to copy file(s) from a module’s agi-bin dir:
chmod(): Operation not permitted
<<

What file should I check ?

31

please, I need someone to comment on this. it should be an easy problem. Just a little experience is required.

32

Pleading is not going to help. I have done all that can be done with the information you have provided.

I already explained why you are not getting responses from anyone else.

33

Unfortunately, we are wrapped up in our own issues (many of which are not related to “older” versions). From a tadpole’s perspective, is there a reason you didn’t install the newest Distro version? You may find a broader audience if you did.

34

is there no one to comment/help ?

35

I don’t know what to say. you guys are really not helping.
Anyway, I found my way out. compiled asterisk-addons from scratch and the cdr_addon module was renewed.

everything working now.

THANKS FOR NOTHING.

I hope I don’t have to come back to this forum again.

36

=========================
On April 7th, 2012 mrmrmrmr (tadpole) said:

I don’t know what to say. you guys are really not helping.
Anyway, I found my way out. compiled asterisk-addons from scratch and the cdr_addon module was renewed.

everything working now.

THANKS FOR NOTHING.

I hope I don’t have to come back to this forum again.

WOW. With that attitude I am sure you will be deeply missed.

37

What a putz. I reread the thread. Nowhere did he mention how he installed any of this. It turned out not to be a FreePBX but an Asterisk issue? It could not have been 1.8 because add-ons are included by default.

Anyone reading this thread, this is a great example of how to alienate an entire community.

38

Wow

A quick google shows Asterisk 1.8.4 has a issue like this. Wonder if he was using 1.8.4. Wish I would of read this sooner.

39

hahaha. I should of looked at his SIP trace. Asterisk FPBX-2.9.0(1.8.4.4)

40

Good to see that at least you understand sometimes looking at a trace would give a clue.
Unfortunately SkykingOH didn’t even try…

I don’t know why Ubuntu 11 repositories have Asterisk 1.8.4.something by default.
But someone with this knowledge (tonyclewis) would have saved me a lot of trouble.

hahaha. at least someone could have checked that trace. SkykingOH requested lots of unavailable logs but didn’t even check them.