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 <[email protected]> Subject: [asterisk-users] asterisk and cisco 2800 To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Message-ID: <[email protected]> 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 [6...@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 [6...@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
