SOLVED two Asterisk PBX servers connected using an IAX2 trunk

voipgkavek

Member
Joined
Nov 26, 2007
Messages
186
Reaction score
5
I had two Asterisk PBX servers connected using an IAX2 trunk. I configured both sides using the instructions from "PBX in a Flash for Newbies - Connecting two Asterisk PBX servers using an IAX2 trunk" http://www.cadvision.com/blanchas/Asterisk/IAX2Trunk.html
It worked beautifully. The machines had the following config
machine A
Code:
PBX in a Flash Version   = 1.7.5.5                     
FreePBX Version          = 2.8.0.4                     
Running Asterisk Version = Asterisk 1.8.0              
Asterisk Source Version  = 1.8.0                       
Dahdi Source Version     = 2.4.0+2.4.0
machine B
Code:
PBX in a Flash Version   = 1.7.5.5          
FreePBX Version          = 2.8.0.4          
Running Asterisk Version = Asterisk 1.8.2.1 
Asterisk Source Version  = 1.8.2.1          
Dahdi Source Version     = 2.4.0+2.4.0


Now I had to reinstall a machine A from scratch, so now
it has
Code:
PBX in a Flash Version = 1.7.5.5
FreePBX Version = 2.8.1.4
Running Asterisk Version = Asterisk 1.8.3.2
Asterisk Source Version = 1.8.3.2
Dahdi Source Version = 2.4.1+2.4.1
machine B stayed the same as it was originally.

the settings are identical to what was in it before (i used freepbx backup and restore and it worked, ive checked manually, and all seems to be exactly the same).

None-the-less I cant make calls between the servers anymore. The firewalls haven't changed, and the 2 servers still register with each other as can be seen here

Code:
Name/Username    Host                 Mask             Port          Status    
Name/Username    Host                 Mask             Port          Status    
SD2TJ-VPN/SdUse  192.168.3.99    (S)  255.255.255.255  4569 (T)      OK (597 ms)
and
Code:
Name/Username    Host                 Mask             Port          Status    
TJ2SD-VPN/TjUse  192.168.2.99    (S)  255.255.255.255  4569 (T)      OK (611 ms)

but when i try to make a call, i get a all circuits are busy message...
this is the CLI output calling from server A->B
Code:
   -- Executing [s@macro-dialout-trunk:19] Dial("SIP/7036-00000011", "IAX2/SD2TJ-VPN/7601,300,tr") in new stack
    -- Called SD2TJ-VPN/7601
    -- Hungup 'IAX2/SD2TJ-VPN-17682'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/7036-00000011", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 58") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/7036-00000011", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)

the result is exactly the same when calling from B->A

Any ideas? could this be because of the upgrade to 1.8.3.2 ?

thanks,
-gk
 
actually in fpbx gui interface you can add it under tools Asterisk IAX settingsnear the bottom of the page you have to put both of the entries in.

Be warned there does seem to be some problems with these settings in asterisk 1.8.x and using different versions of asterisk 1.8.x on different machines. I have one pair that uses older 1.8.x and one box is current. Couldn't make IAX2 work at all. Eventually I ran update source on the remote side and with both boxes the same version the iax2 trunk seems to work.

Eventually I set up a simple sip trunk thru the vpn which proved to be adequate.


Really since they "improved" security for iax2 it has not worked as reliably as before.


Such is the price of progress.


Tom
 
Thanks Tom and Atsak,
Here is were we stand. I tried adding

-----------
calltokenoptional = 0.0.0.0/0.0.0.0
maxcallnumbers = 16382

to /etc/asterisk/iax_general_custom.conf
-----------
on both sides, but it made no difference. Since I used blanchae's spreadsheet to create the trunks, both sides already had calltokenrequired=no

Finally, calling from server A to server B this is what i see on the log of B (the receiving side) (server A says chanel unavailable)

--------------------------
[2011-04-04 19:08:39] NOTICE[9676] chan_iax2.c: Rejected connect attempt from 192.168.2.99,
requested/capability '0x400000000000000 (nothing)'/'0xffff7f0c03800000 (nothing)' incompatible
with our capability '0xe (gsm|ulaw|alaw)'.
-------------------------

I added an
allow=all

on both sides for testing but i got the same error on both sides.
Anything else I can try? suggestions?

thanks,
-gk
 
Did you restart asterisk with amportal restart?
 
--------------------------
[2011-04-04 19:08:39] NOTICE[9676] chan_iax2.c: Rejected connect attempt from 192.168.2.99,
requested/capability '0x400000000000000 (nothing)'/'0xffff7f0c03800000 (nothing)' incompatible
with our capability '0xe (gsm|ulaw|alaw)'.
-------------------------

I added an
allow=all

on both sides for testing but i got the same error on both sides.
Anything else I can try? suggestions?

thanks,
-gk

Try
disallow=all
followed by
allow=ulaw&alaw&gsm

See if that does anything?
 
I have already tried

disallow=all
allow=ulaw

on both sides. It made no difference.
I will try what you suggest when I get a chance to restart services on the machines.

thanks,
-gk
 
Hi,
I feel your pain. We've all been there one time or another. But from reading all the posts in this thread, I feel the problem might be in your configuration. If the call is being rejected by server "B", it is safe to assume the you're not authenticating properly. Could you post your config. for one or both servers? I'm almost certain the problem would be solved from there.

Robin
 
I have the same issue, was this solved? The only difference with the trunk on my side that it keeps on saying "Everyone is busy/congested at this time (1:0/0/1)". I have other machines which is both asterisk 1.8, i have no issues with them interconnecting, only when interconnecting with asterisk 1.4 to asterisk 1.8.
 
I had this same issue, and although I didn't touch my firewall settings (I blame the CEO), there was indeed a typo in the firewall policy dealing with this IAX2 situation that was blocking their communication.
 

Members online

Forum statistics

Threads
26,687
Messages
174,409
Members
20,257
Latest member
Dempan
Get 3CX - Absolutely Free!

Link up your team and customers Phone System Live Chat Video Conferencing

Hosted or Self-managed. Up to 10 users free forever. No credit card. Try risk free.

3CX
A 3CX Account with that email already exists. You will be redirected to the Customer Portal to sign in or reset your password if you've forgotten it.
Back
Top