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

[Jonathan Thurman]

We are in a similar situation to you as far as moving from Cisco to
Asterisk.  I have not got to the point of integrating Asterisk
directly to our PSTN gateways yet, but this might help.

On our H.323 gateways we use trunk groups for outbound call hunting.
You can create a single trunk group for outbound calls and add any
voice port to it that you want.  On the voice port you tell it what
trunk group it belongs to and the priority (1-64 I believe).  The
available port with the lowest priority wins.  This configuration is
from a 2801 using H.323 to CallManager 6.1:

Example:
-------------------------

trunk group Outbound
 description - Outbound calling hunt group
 hunt-scheme sequential
!

voice-port 0/0
 trunk-group Outbound 1
!
voice-port 2/0
 trunk-group Outbound 2
!

dial-peer voice 2000 pots
 trunkgroup Outbound
 description Outbound call hunting
 destination-pattern .T
!

------------------

-Jonathan


On Thu, Apr 23, 2009 at 12:18 PM, Jimmy Ezell <jezell at hmhca.com> wrote:
> 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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