[asterisk-users] asterisk and cisco 2800

Peder peder at networkoblivion.com
Tue Jul 6 07:04:10 CDT 2010


That's not right.  Should be 1245 -> 4512:

http://www.voip-info.org/wiki/view/crossover+T1+cable



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Tuesday, July 06, 2010 2:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk and cisco 2800

Hi Neeraj,

my problem is not terminating but making the Cisco accept the calls 
coming from my Asterisk. The bad news is I cannot have access to the 
Cisco sw, it is like a black box for me. The only thing I can have 
access to is the T1/E1 port on the back of the Cisco 2800.
I made a custom cable too:

1 <--> 5
2 <--> 4
4 <--> 2
5 <--> 1

and it seems to work because I get all alarms off after plugging it in.

Thank you

Giorgio Incantalupo


Neeraj Chand wrote:
> Hi Giorgio, 
>
> Why don't you terminate calls on the cisco router via SIP? 
>
>
>
> ------------------------------
>
> Message: 11
> Date: Fri, 02 Jul 2010 18:54:31 +0200
> From: Giorgio Incantalupo <gincantalupo at fgasoftware.com>
> Subject: [asterisk-users] asterisk and cisco 2800
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 	<asterisk-users at lists.digium.com>
> Message-ID: <4C2E19C7.5090909 at fgasoftware.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Hi all,
>
> I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures
>
> with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the 
> cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives 
> no errros, the span is up and active, green light on the card) but when 
> I make a test with my iax phone, there's no way to dial the PBX and I 
> get this WARNING:
>
> [Jul  2 15:20:36] VERBOSE[15004] logger.c:     -- Accepting 
> AUTHENTICATED call from XXX.XXX.XXX.XXX:
>        > requested format = gsm,
>        > requested prefs = (),
>        > actual format = gsm,
>        > host prefs = (),
>        > priority = mine
> [Jul  2 15:20:36] VERBOSE[15031] logger.c:     -- Executing 
> [6666 at inbound:1] Dial("IAX2/1-1024", "DAHDI/g2/XXXXXXXXX|60|tT") in new 
> stack
> [Jul  2 15:20:36] WARNING[15031] app_dial.c: Unable to create channel of
>
> type 'DAHDI' (cause 0 - Unknown)
> [Jul  2 15:20:36] VERBOSE[15031] logger.c:   == Everyone is 
> busy/congested at this time (1:0/0/1)
> [Jul  2 15:20:36] VERBOSE[15031] logger.c:     -- Executing 
> [6666 at inbound:2] Hangup("IAX2/1-1024", "") in new stack
> [Jul  2 15:20:36] VERBOSE[15031] logger.c:   == Spawn extension 
> (inbound, 6666, 2) exited non-zero on 'IAX2/1-1024'
> [Jul  2 15:20:36] VERBOSE[15031] logger.c:     -- Hungup 'IAX2/1-1024'
>
> Any hints?
>
> Thank you.
>
> Giorgio Incantalupo
>
>
>
>
>
>   


-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list