Hi Peder, I'make a new cable following the info on that webpage. I hope it works with Cisco 2800 too! :)
Thank you! Giorgio Incantalupo Peder wrote: > That's not right. Should be 1245 -> 4512: > > http://www.voip-info.org/wiki/view/crossover+T1+cable > > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@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 <gincantal...@fgasoftware.com> >> Subject: [asterisk-users] asterisk and cisco 2800 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <asterisk-users@lists.digium.com> >> Message-ID: <4c2e19c7.5090...@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 >> [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