[wplug] Installfest Aftermath - Asterisk Progress

Greg Simkins gregsim at telerama.com
Thu May 26 16:25:30 EDT 2005


Yes, the TDM400P has space for four daughter cards and was shipped with one 
FXS module and one FXO module.  (Actually my face is red - I hadn't really 
paid much attention when I installed the card, but believed the photograph 
that came with the card showing all four modules installed - so I only have 
ports 1 and 4 active).

Thanks for the advice, Scott.

----- Original Message ----- 
From: "smk" <smk at fyi.net>
To: "General user list" <wplug at wplug.org>
Sent: Thursday, May 26, 2005 4:06 PM
Subject: Re: [wplug] Installfest Aftermath - Asterisk Progress


>
> Greg,
>
> I assume that card is a FXS card then? (question I had during the install
> fest)
>
> If you are planning on using a SIP provider port 5060 should be fine.  I
> am using a linksys and it worked fine at my place after forwarding the
> port.  I will double check my config when I get home and let you know if
> there was anymore ports to forward.
>
> You may also surf around www.voip-info.org and see what you find there?
>
> Scott
>
>
>
>
>
> On Thu, 26 May 2005, Greg Simkins wrote:
>
>> I am looking for some help from any Asterisk users on our list.
>>
>> Last saturday at the installfest, I started installation of Asterisk on a 
>> Suse Linux distro.  Thanks to all those that helped me struggle through 
>> the instructions that came with the Digium TDP100P card.  With help from 
>> several WPLUGers, I made tremendous progress, but I am not yet making 
>> phone calls.
>>
>> With some additional help from Digium support, I now have a dial tone. 
>> (It turns out that Suse included a bunch of drivers for the Digium card 
>> in their standard install.  In their attempt to be helpful, they ended up 
>> torpedoing me, since their revisions conflicted with the current version 
>> of Asterisk, even though it is the recently issued ver 9.2 of Suse.  The 
>> old drivers all had to be deleted).
>>
>> I am not yet able to dial out or receive calls. I included all the 
>> instructions from Asterlink in the sip.conf, iax.conf and extensions.conf 
>> files.  (3 of the dozen or so configuration files that Asterisk uses).
>>
>> In listening to Ryan's presentation on the WPLUG web site, I wonder if I 
>> may have a problem with my Linksys router.  I forwarded port 22 to my 
>> Asterisk server so I am able to get a command line remotely.  It appears 
>> that SIP uses port 5060, so I will open that port tonight when I get 
>> home.
>>
>> Are there any other ports I should open on the firewall?  Is there 
>> something in the dialplan that I need to fix other than what Asterlink 
>> provided?
>>
>> Thanks,
>>
>> Greg
>
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