Hi Giorgio,
Why don't you terminate calls on the cisco router via SIP?
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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