[asterisk-users] SIP interconnection problem

[Robert Bielik]

Tarek Sawah skrev:
> you need to post you SIP.conf and your Extensions.conf so someone can 
> have a look at them and see if there is anything missing
> what are the contexts you are using with your peers?
> what is the dial plan triggered when calling your destination number?

Machine 1 -------------------------------------------------------
iax.conf: ======================
[general]
bandwidth=low
disallow=lpc10                  ; Icky sound quality...  Mr. Roboto.
jitterbuffer=no
forcejitterbuffer=no
autokill=yes

[2200]
type=friend
host=dynamic
context=users
username=2200
secret=none
auth=md5

sip.conf =======================
[general]
port=5060
bindaddr=0.0.0.0

disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726

dtmfmode=rfc2833

register => machine_1:wabooba at 192.168.10.77/machine_2

[machine_2]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_2

extensions.conf ==================
[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.77
OUTGOING_PREFIX=0

[default]
include => sip-incoming
include => test

[test]
; Create an extension, 600, for evaluating echo latency.
;
exten => 600,1,Playback(demo-echotest)  ; Let them know what's going on
exten => 600,n,Echo                     ; Do the echo test
exten => 600,n,Playback(demo-echodone)  ; Let them know it's over
exten => 600,n,Goto(s,6)

[users]
include => sip-incoming
include => outgoing
include => test

[sip-incoming]
include => agi-async
include => internal

[agi-async]
exten => _01XXXX,1,Agi(agi:async)

[internal]
exten => _2XXX,1,NoOp()
exten => _2XXX,n,Dial(IAX2/${EXTEN})
exten => _2XXX,n,Hangup()

[outgoing-agi-async]
exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${EXTEN}@${SIP_TRUNK})
exten => _${OUTGOING_PREFIX}.,n,Set(CALLERID(name)=reason-${DIALSTATUS})
exten => _${OUTGOING_PREFIX}.,n,Agi(agi:async)

[outgoing]
exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${SIP_TRUNK}/${EXTEN:1})
exten => _${OUTGOING_PREFIX}.,n,Hangup()

Machine 2 --------------------------------------------------------
sip.conf =======================
[general]
port=5060
bindaddr=0.0.0.0

disallow=all
allow=alaw ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=g726

dtmfmode=rfc2833

register => machine_2:wabooba at 192.168.10.11/machine_1

[machine_1]
allow=alaw,ulaw,gsm,g726
host=dynamic
secret=wabooba
type=friend
context=sip_incoming
username=machine_1

extensions.conf ==================
[globals]
; The outgoing sip trunk
SIP_TRUNK=192.168.10.11

Rest is exactly the same. I have a zoiper connected to each machine and I'm trying to make a call from Machine 2 to zoiper
on Machine 1:

    -- Registered IAX2 '2200' (AUTHENTICATED) at 192.168.10.113:4569
    -- Accepting AUTHENTICATED call from 192.168.10.113:
       > requested format = gsm,
       > requested prefs = (),
       > actual format = gsm,
       > host prefs = (),
       > priority = mine
    -- Executing [02200 at users:1] Dial("IAX2/2200-1200", "SIP/192.168.10.11/2200") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 192.168.10.11/2200
[Oct 26 09:20:25] NOTICE[20248]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '"2200" <sip:2200 at 192.168.10.77>;tag=as6173091f'
    -- SIP/192.168.10.11-090c2ea8 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [02200 at users:2] Hangup("IAX2/2200-1200", "") in new stack
  == Spawn extension (users, 02200, 2) exited non-zero on 'IAX2/2200-1200'
    -- Hungup 'IAX2/2200-1200'

Besides that "sip show peers" on either machine shows the other one correctly registered, and "iax2 show peers" shows the connected zoiper on each machine.
	
Ideas, please ??

TIA
/Rob