[asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help

[Jimmy Ezell]

Dan thank you, yes that seems to help.  It looks like the bridging is happening now and I see the light come on in the second FXO port, but then I get a busy signal after that and the call still does not complete.  If I set the second line as priority 1 it completes the first call on that line and second call gets the busy on the first line.  I even tried moving the lines to a different FXO card and the result is the same.

Here is my current config for the cisco dial-peers:


dial-peer voice 2212 pots
 preference 2
 destination-pattern .T
 port 2/0
 forward-digits all
!
dial-peer voice 2211 pots
 preference 1
 destination-pattern .T
 port 0/0
 forward-digits all


Thanks again Dan,  I think I am much closer now.
Jimmy




-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Dan Austin
Sent: Thursday, April 23, 2009 09:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Step-by-Step Asterisk and Cisco 1760 Help


Jimmy wrote:

Second Call out the asterisk console looks like this-----------------------------------------------------:
    -- Executing [92952210 at internal:1] Dial("SIP/222-09ab3588", "SIP/Cisco1760/2952210") in new stack
    -- Called Cisco1760/2952210
[Apr 22 16:08:58] NOTICE[3450]: chan_sip.c:14489 handle_request_invite: Call from '222' to extension '2952210' rejected because extension not found.
    -- Got SIP response 486 "Busy here" back from 172.17.2.1
    -- SIP/Cisco1760-09ab7cf8 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [92952210 at internal:2] Congestion("SIP/222-09ab3588", "") in new stack
  == Spawn extension (internal, 92952210, 2) exited non-zero on 'SIP/222-09ab3588'
localhost*CLI>


--------------sip.conf ---------
[general]
bindaddr=0.0.0.0

[Cisco1760]
context=incoming_calls
type=friend
host=172.17.2.1
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very


----------extensions.conf------------
[globals]
OUTBOUNDTRUNK=SIP/Cisco1760


[outbound-local]
exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _9NXXXXXX,n,Congestion()
exten => _9NXXXXXX,n,Hangup()

-----------Cisco 1760 config ----------
dial-peer voice 100 pots  (This line that is set to preference 2 does not work)
 huntstop
 preference 2
 destination-pattern .T
 port 0/0
 forward-digits all
!
dial-peer voice 2212 pots    (This line that is set to Preference 1 is the one that works)
 huntstop
 preference 1
 destination-pattern .T
 port 0/1
 forward-digits all


------------------------------------------------
You do not want to use huntstop on the dialpeers in this situation.
The huntstop option tells the call routing function in the router to
stop search for a call route if it encounters a failure.

Call number 2 hits dialpeer 1, finds it busy and the huntstop causes
the processing to stop.

Dan

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