FreePBX & RASPBX trunked IAX2

Good morning all,

6XXX & 7XXX on the raspbx can call extensions on the FreePBX 1XXX & 2XXX
1XXX & 2XXX cannot call the 6XXX or 7XXX

Raspbx running 11 extensions 6XXX 7XXX

FreePBX 13 extensions 1XXX , 2XXX

I setup my trunk and outbound routes on both sides. The trunk is up.

Second Server: RASPBX

Trunk Name: MASTER

Peer details:

host=192.168.15.1
username=Slave1
secret=XXXXXXXXXXXXXX
type=friend
context=from-internal
transfer=no
allow=alaw&ulaw
disallow=all

Free PBX :: running on Pentium i7

Trunk Name: Slave1

Peer details:

host=192.168.15.61
username=Master
secret=XXXXXXXXX
type=friend
context=from-internal
transfer=no
allow=alaw&ulaw
disallow=all

Outbound routes
all routes are intra-comany

On the FreePBX
dial pattern [6]XXX & [7]XXX out Trunk Master

on the RASPBX
dial pattern [1]XXX & [2]XXX out Trunk SLAVE1

Logs (/var/log/asterisk/full) will tell the tale.

Without that, there are only about 1000 possibilities.

You have unintentially disallowed all codecs, order is important.

It appears that as a new person here I can’t post a file or even paste its contents here.

This is a lot of help for a legitimate problem

Dicko,
That has been fixed. I pulled that from a file I had earlier this morning. sorry about that.

Make sure your systems are compatible

https://wiki.asterisk.org/wiki/display/AST/IAX2+Security

I didn’t see anything wrong after reviewing that document.

this is what I see. This is working from the raspbx
Executing [email protected]:23 Dial(SIP/6001-00000010, IAX2/Slave1/1001,300,Ttr
app_dial.c: – Called IAX2/Slave1/1001
chan_iax2.c: – Call accepted by 192.168.15.1 (format alaw)
chan_iax2.c: – Format for call is (alaw)
app_dial.c: – IAX2/Slave1-4778 is ringing
app_dial.c: – IAX2/Slave1-4778 is ringing
chan_iax2.c: – Hungup IAX2/Slave1-4778

This is dial from the FreePBX PC to the raspbx
pbx.c: – Executing [[email protected]:17] Macro( SIP/1001-00000021, dialout-trunk-predial-hook, in new stack
pbx.c: – Executing [[email protected]:1] MacroExit( SIP/1001-00000021 ) in new stack
pbx.c: – Executing [[email protected]:18] GotoIf ( SIP/1001-00000021 , 0?bypass,1) in new stack
pbx.c: – Executing [[email protected]:19] ExecIf ( SIP/1001-00000021, “1?Set(CONNECTEDLINE(num,i)=6001)”) in new stack
pbx.c: – Executing [[email protected]:20] ExecIf(“SIP/1001-00000021”, “1?Set(CONNECTEDLINE(name,i)=CID:1001)”) in new stack
pbx.c: – Executing [[email protected]:21] ExecIf(“SIP/1001-00000021”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)1001)”) in new stack
pbx.c: – Executing [s AT macro-dialout-trunk:22] GotoIf(“SIP/1001-00000021”, “0?customtrunk”) in new stack
pbx.c: – Executing [s AT macro-dialout-trunk:23] Dial(“SIP/1001-00000021”, “IAX2/Master/6001,300,Ttr”) in new stack
app_dial.c: – Called IAX2/Master/6001
chan_iax2.c: – Hungup 'IAX2/Master-4876’
app_macro.c: == Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on ‘SIP/1001-00000021’ in macro 'dialout-trunk’
pbx.c: == Spawn extension (from-internal, 6001, 7) exited non-zero on 'SIP/1001-00000021’
pbx.c: – Executing [[email protected]:1] Macro(“SIP/1001-00000021”, “hangupcall”) in new stack

I suggest you look in your /var/log/asterisk/full files on both machines and match the timestamps at the beginning of each for a cross correlation of

app_dial.c: – Called IAX2/Master/6001
chan_iax2.c: – Hungup ‘IAX2/Master-4876’

Hi Dicko,
Well I found a typo and thats what the issue was. That one last little bug … Thank you for your suggestions earlier today. Hope you have a great weekend…

Mike