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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: domainwizard on February 24, 2008, 07:19:32 PM
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i am routing all outbound traffic thru a primary sail box for accounting purposes. The sail to sail trunks are perfect and function flawlessly.
I have one freepbx box that i have tried to connect, trunking to the sark box over an internal lan
on sark i have this
[hellocallme]
type=peer
host=192.168.xxx.xxx
qualify=3000
canreinvite=no
username=hellocallme
secret=asterisk
trunk=yes
disallow=all
allow=alaw
allow=ulaw
[callmehello]
type=user
context=internal
secret=asterisk
on freepbx i have this .
[callmehello]
allow=alaw&ulaw
canreinvite=no
disallow=all
host=192.168.xxx.xxx
qualify=3000
secret=asterisk
trunk=yes
type=peer
username=hellocallme
[callmehello]
context=internal
secret=asterisk
type=user
the trunk seems to be happily connected, but if i initiate a call from the freepbx box, i get " call declined" and the following error in cli...
"chan_iax2.c:7241 socket_process: Rejected connect attempt from 192.168.xxx.xxx, who was trying to reach 'NXXNXXXXXX@' "
of course when i initiate a call from another sark box, all is well
many regards
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Hi there
In the end, they're both just asterisk boxes so what you are attempting should work. One thing confuses me...
who was trying to reach 'NXXNXXXXXX
Is that the exact message or have you substituted the NXXNXXXXXX for a real number?
Can you clarify please?
Also, I'm assuming that your goal is to have the FreePBX server use the SAIL trunks.. yes?
Kind Regards
S
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yes, that is my goal and yes, i substituted NXX for the correct outbound number, however the ,after @ is exact
the paragraph from verset describes this as a bug
http://forums.digium.com/viewtopic.php?p=45689&sid=59387365a0502a19df07c4af2c78801e
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[hellocallme]
type=peer
host=192.168.xxx.xxx
qualify=3000
canreinvite=no
username=hellocallme
secret=asterisk
trunk=yes
disallow=all
allow=alaw
allow=ulaw
[callmehello]
type=user
context=internal
secret=asterisk
on freepbx i have this .
[callmehello] <=======================
allow=alaw&ulaw
canreinvite=no
disallow=all
host=192.168.xxx.xxx
qualify=3000
secret=asterisk
trunk=yes
type=peer
username=hellocallme
[callmehello] <=======================
context=internal
secret=asterisk
type=user
There's a small error here but I'm not sure it's the whole story (should be hellocallme in the freepbx user). Can you show us Freepbx's dial statement for this trunk please? - turn on iax debugging (iax2 set debug) and then run the call. let me see the console output. (iax2 set debug off) to turn it off again.
Kind Regards
S
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1800NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX
1NXXNXXXXXX
800NXXXXXX
866NXXXXXX
877NXXXXXX
888NXXXXXX
NXXNXXXXXX
NXXXXXX
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00012ms SCall: 16384 DCall: 00000 [192.168.111.12:4569]
VERSION : 2
CALLED NUMBER : 2821949
CODEC_PREFS : (alaw|ulaw)
CALLING NUMBER : 5000
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : bc
LANGUAGE : en
USERNAME : hellocallme
FORMAT : 4
CAPABILITY : 57356
ADSICPE : 2
DATE TIME : 2008-02-24 16:27:14
[Feb 24 15:27:50] NOTICE[4777]: chan_iax2.c:7241 socket_process: Rejected connect attempt from 192.168.111.12, who was trying to reach '282XXX1949@'
Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00012ms SCall: 00002 DCall: 16384 [192.168.111.12:4569]
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT
Timestamp: 00019ms SCall: 00002 DCall: 16384 [192.168.111.12:4569]
CAUSE : No authority found
CAUSE CODE : 50
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CALLED NUMBER : 2821949
CODEC_PREFS : (alaw|ulaw)
CALLING NUMBER : 5000
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : bc
LANGUAGE : en
USERNAME : hellocallme <================================
FORMAT : 4
CAPABILITY : 57356
ADSICPE : 2
DATE TIME : 2008-02-24 16:27:14
I can see what's wrong. I' not sure how you fix it but maybe you will when I explain. You are driving the call into SARK with a user of "hellocallme". The SARK user you have set-up to receive the call is called "callmehello". This is why you are getting the "No Authority Found" message. Asterisk can't fins an entry in IAX.CONF to match.
Hope this helps.
Kind Regards
S