[asterisk-users] asterisk and cisco 2800

[Peder]

If you do back to back, then one end needs to clock.  To set it on the
Cisco, type "clock source internal" under the controller config.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, July 09, 2010 4:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk and cisco 2800

Hi Peder,

it seems to work, thank you!

Now I've got a problem with the cisco 2800 which is resetting every 5 
minutes but I do not think it is related to the cable, maybe something 
about the clock but except for a wiki page 
(http://www.voip-info.org/wiki/view/Asterisk+legacy+integration) there 
is nothing on internet about connecting asterisk and cisco... :(

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-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
>>
>>
>>
>>
>>
>>   
>>     
>
>
>   


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