..all circuits are busy now, please try again later

New to VOIP, please be indulgent…
So, I purchased a line from Axiatel, and had a professional consultant help me configure the server.
The iso was installed to a VM.
Some time the calls go out properly, and the quality is very good, but MOST of the times I just get “…all circuits are busy now, please try again later”.
My consultant says it’s the other end refusing the connection, and I must trust hime because his is an Asterisk specialist, and malkes a living from it. Axiatel on the other side says everythink is working 100%, and I must trust them because if all their customers experienced this down rate, they would close business in one week.
So, as usual, I’m stuck in the middle.
Any advice from the experts out there?

What is your IT skill level? Are you able to capture logs and call traces when calls don’t work?

More than likely it’s your firewall.

If you get the message and try again does it almost always work on 2nd or 3rd try?

thanks for your kind reply.

Yes, I can cat & grep logfiles (with a little help from my friends…)

The funny thing is that this error is not consistent but random, and occurs most of the time.

My firewall is an Ipfire distro.
Freepbx is installed from ISO on a VM.

What should I look for on the firewall?

ops, I forgot to tell you that retrying doesn’t work.

-G

What SIP cause code are you getting when it fails. Your asterisk logs should show you

Tony,

thanks for your kind reply.

I don’t know if this is the relevant part of the log:

[2014-07-25 12:19:44] VERBOSE[16748][C-00000094] netsock2.c:   == Using SIP RTP TOS bits 184
[2014-07-25 12:19:44] VERBOSE[16748][C-00000094] netsock2.c:   == Using SIP RTP CoS mark 5
[2014-07-25 12:19:44] VERBOSE[16748][C-00000094] app_dial.c:     – Called SIP/34181/00234535167
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] app_dial.c:   == Everyone is busy/congested at this time (1:0/0/1)
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:23] NoOp(“SIP/12-00000105”, "Dial failed for some rea$
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:24] GotoIf(“SIP/12-00000105”, "0?continue,1:s-CHANUNA$
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:1] Set(“SIP/12-00000105”, “RC=1”) in new $
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:2] Goto(“SIP/12-00000105”, “1,1”) in new $
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Goto (macro-dialout-trunk,1,1)
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:1] Goto(“SIP/12-00000105”, “continue,1”) in new stack
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Goto (macro-dialout-trunk,continue,1)
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:1] NoOp(“SIP/12-00000105”, "TRUNK Dial failed $
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:2] Set(“SIP/12-00000105”, "CALLERID(number)=12$
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:7] Macro(“SIP/12-00000105”, “outisbusy,”) in new$
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:1] Progress(“SIP/12-00000105”, “”) in new stack
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:2] GotoIf(“SIP/12-00000105”, “0?emergency,1”) in new stack
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:3] GotoIf(“SIP/12-00000105”, “0?intracompany,1”) in new s$
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:4] Playback(“SIP/12-00000105”, "all-circuits-busy-now&pls$
[2014-07-25 12:19:45] VERBOSE[16748][C-00000094] file.c:     – <SIP/12-00000105> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
[2014-07-25 12:19:47] VERBOSE[16748][C-00000094] file.c:     – <SIP/12-00000105> Playing ‘pls-try-call-later.ulaw’ (language ‘en’)
[2014-07-25 12:19:48] VERBOSE[16748][C-00000094] app_macro.c:   == Spawn extension (macro-outisbusy, s, 4) exited non-zero on ‘SIP/12-00000105’ in ma$
[2014-07-25 12:19:48] VERBOSE[16748][C-00000094] pbx.c:   == Spawn extension (from-savona-uff, 0234535167, 7) exited non-zero on ‘SIP/12-00000105’
[2014-07-25 12:19:48] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:1] Macro(“SIP/12-00000105”, “hangupcall,”) in new stack
[2014-07-25 12:19:48] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:1] GotoIf(“SIP/12-00000105”, “1?theend”) in new stack
[2014-07-25 12:19:48] VERBOSE[16748][C-00000094] pbx.c:     – Goto (macro-hangupcall,s,3)
[2014-07-25 12:19:48] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:3] ExecIf(“SIP/12-00000105”, "0?Set(CDR(recordingfile)=)$
[2014-07-25 12:19:48] VERBOSE[16748][C-00000094] pbx.c:     – Executing [[email protected]:4] Hangup(“SIP/12-00000105”, “”) in new stack

In your case I would look at your line use routing and Voice mail.
With bigger systems I believe pre-processor speed or internet traffic can cause this problem
I vaguely remember having this issue but recall getting the error when i dialed too fast or switched
lines to quickly.
Take this with a grain of salt.
I would ask dicko
…all circuits are busy now, please try again later

Also try disabling voice mail and forwarding to see if the problem goes away

starting to remember when i had this issue.
Please take a look at how many trunks are registered and how many lines in use.

At the main page there are 3 status bars. trunks, trunks online and i believe lines in use.
My system only works with one trunk registered.
I have no idea why.
The sequence in which your router obtains ip address adjacent to your phone and computer
was my issue.
Try booting your phone then reset your router then get ip for pc
in other words its not 1-2-3 but rather 2-3-1

I have some more info on this issue.

I thought to register a phone directly with Axiatel, turning off the Freepbx server.
It works like a charm, local/foreign/mobile phones reached with 100% success.
And it goes through the firewall too, so I think we can rule out both the firewall and the provider.

This gives me hopes I will be able setup the Freepbx …someday.

Unfortunately i cant give you anymore advice as i don’t know enough but
can say it took me over 8 months at approx 450 hours to figure mine out.
it was rewarding and both headache at the same time.
Until someone in this forum told me my router was causing the problem.
He was right the 550 D-link caused my main issue.
I don’t think this is your issue but expand your search a little to other forums
and try to Quentin your main target to eliminate your problem.
Don’t mess with any thing else but your target.
consider getting a cheap hard drive and uploading the latest stable version.
once working don’t make any major changes or upload modules that may
cause problems until you can back your system up
with a second hard drive.

miss spelled Quarantine

Ok, I think I made it.
It’s been 9 hrs now, and not a single “all circuits are busy”.
I post here my solution, for public benefit.
Google hinted that the problem could lie in the peer detail of the trunk, when fromuser is missing.
It was missing indeed, and I also noticed that the userid was entered in the fromdomain field, instead of a domain or IP.
Too bad I purchased one hr of paid support and 50 min have gone without any real help.

Trunk settings are provider specific. A support technician has no way to keep track of details for each and every single trunk out there. This is why that area is a textbox and not individual settings, because in the greater concept of things there are many different combinations of what can go in there.

Provider specific is the combination of userid/secret/ISPdomain.

The rest is a matter of FreePbx: if it wasn’t, why bother describing the entries in the Wiki?

@glsarto it’s unfortunate that you are not happy with the level of support provided, however I think you are under some wrong assumptions about SIP, the Session Initiation Protocol or SIP standard takes close to 300 pages to define, if it was always as simple as entering three settings then that standard would be a much shorter document, and FreePBX would not provide you with additional options that may be required depending on how your specific provider sets up their network, and provisions your service. These requirements range pretty widely just in the providers that are known to support open SIP standards and FreePBX. When you start dealing with carriers that do not provide support, or are unwilling to assist you with your configuration if you do not purchase their hardware (as noted in your support ticket), then having any third party assist with setup, whether they are FreePBX experts or not, is going to require both some research time and testing.

ok, boys, I give up! don’t shoot!

I’m just that lucky guy that, knowing nothing about VOIP, and having read not a single page of 300 defining SIP standards, picked up the right parameter and was able to fix the dreaded “all circuits are busy” issue… too bad I didn’t buy that lottery ticket, the other morning…

Kidding aside, I did not want to be polemic: quality systems rely on customer satisfaction, and I always welcome my customers’ opininon, whether it is good or not.

So, because of CEO’s contribution to this thread, I thought that the Company did care about customers, and would make some use of my comments.

This time I was not not lucky, really… I purchased 10 hrs of support (750 Euro) from a local professional, who set up the trunk in a way it didn’t work properly, and 1 hr of support from Schmooze…

But thanks Google, I made it!

BTW, would you be interested in receiving my CV?

Just kidding…

have a nice day,

:slight_smile:

-GianLuca

I do post here all the time and reviewed your ticket. I am sorry we were not able to resolve your issue but had your provider given us real settings that we could translate into FreePBX/Asterisk things would of gone smoother. One of the hardest thing in Asterisk is dealing with SIP carriers who do not provide example configs for their setup as there are hundreds of options and I am sure you will find other issues as you go along such as Caller ID not working or things like Follow Me not sending the correct Caller ID as Caller ID itself can be handled numerous ways and combination of ways and each provider is different.