asterisk users - Jun 2005 - Reccomendations for connecting to 3-4 PSTN lines?

Hello,

   I'm looking to connect Asterisk with three (four in the future) PSTN
lines, and would like to get some opinions on the TDM400 Digium card, vs.
sip gateways like the Mediatrix 1204, vs. other hardware solutions I'm not
yet aware of.

   I need the ability to prioritize which PSTN lines are used for outgoing
calls (I understand this can be done with the Mediatrix --
http://lists.digium.com/pipermail/asterisk-users/2004-February/035926.html),
and only answer two of the three lines (the third line will be answered by
a directly connected FAX machine, but we can use that line for outgoing
calls if the first two PSTNs are in use).

   Does the higher cost of something like the Mediatrix ($634) bring
advantages to the table over the TDM04b ($337)?  Does the TDM card provide
functionality which the sip gateways do not (e.g. incoming / outgoing call
progress)?

I appreciate hearing from folks who have experience with both methods of
connectivity,

Thanks - Ivan.

In my experience if you can get an external device to do the PSTN <>
SIP for you, like an ATA, then it works much better than the Digium
TDM400 cards. If you don't mind the extra ~$200 just go for it. The
other option might be a channel bank (like the ADIT 600) and a Digium
T1 card. This will let you use up to 24 FXO/FXS per 8 lines to a card
(they can be mixed FXS and FXO), but again this will have Asterisk do
the transcoding (if any) from ulaw to 729 or whatever, which is usualy
much better to do before it reaches Asterisk.

On 6/1/05, Ivan Fetch <voip@ivanfetch.com> wrote:
> Hello,
> 
>   I'm looking to connect Asterisk with three (four in the future) PSTN
> lines, and would like to get some opinions on the TDM400 Digium card, vs.
> sip gateways like the Mediatrix 1204, vs. other hardware solutions I'm
not
> yet aware of.
> 
>   I need the ability to prioritize which PSTN lines are used for outgoing
> calls (I understand this can be done with the Mediatrix --
>
http://lists.digium.com/pipermail/asterisk-users/2004-February/035926.html),
> and only answer two of the three lines (the third line will be answered by
> a directly connected FAX machine, but we can use that line for outgoing
> calls if the first two PSTNs are in use).
> 
>   Does the higher cost of something like the Mediatrix ($634) bring
> advantages to the table over the TDM04b ($337)?  Does the TDM card provide
> functionality which the sip gateways do not (e.g. incoming / outgoing call
> progress)?
> 
> I appreciate hearing from folks who have experience with both methods of
> connectivity,
> 
> Thanks - Ivan.
> 
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>

>    I'm looking to connect Asterisk with three (four in the future) PSTN
> lines, and would like to get some opinions on the TDM400 Digium card, vs.
> sip gateways like the Mediatrix 1204, vs. other hardware solutions I'm
not
> yet aware of.
> 
>    I need the ability to prioritize which PSTN lines are used for outgoing
> calls (I understand this can be done with the Mediatrix --
>
http://lists.digium.com/pipermail/asterisk-users/2004-February/035926.html),
> and only answer two of the three lines (the third line will be answered by
> a directly connected FAX machine, but we can use that line for outgoing
> calls if the first two PSTNs are in use).
> 
>    Does the higher cost of something like the Mediatrix ($634) bring
> advantages to the table over the TDM04b ($337)?  Does the TDM card provide
> functionality which the sip gateways do not (e.g. incoming / outgoing call
> progress)?
> 
> I appreciate hearing from folks who have experience with both methods of
> connectivity,
I wrote the posting referred to above and its been a while since I've
touched the 1204, but here are a couple of comments based on the
firmware that was available at that time.

Mediatrix 1204:
 - box was designed to work with the 1104 as a toll-bypass solution,
   therefore it does not implement many sip protocol functions
 - it functioned very well, good echo cancellation, audio quality, etc.
 - can only be configured via snmp (no telnet, no web). Their configuration
   tool _must_ match the version of firmware, therefore for each firmware
   upgrade you have to remove and reinstall a new config tool (Win32 based)
 - their snmp implementation had no security, therefore one should not
   expose the box to the internet (eg, hacking)
 - each firmware upgrade is a chargable item (not free) and support is
   basically limited to whatever the reseller can provide (usually not
   much help). Very limited help on this list as there aren't very many
   of these implemented with asterisk.
 - it does not implement the sip "register" function, therefore no way
   to know when the box might be unreachable (for whatever reason)
 - documentation sucks unless you are very familiar with reading snmp
   mib variables, translating their language into telephony and * words,
   etc. Very steap learning curve, but once working, pretty solid box.
 - a little company research (in 2004) suggested the company was having
   financial problems, therefore the only valid assumption one can make
   is they may not be around in the long run.
 - I never did test modem use through the 1204, so no idea if faxing or
   point-of-sale devices will work (or how well).

Digium TDM card:
 - works fairly well for pure voice use, but quality of echo cancellation
   and audio is below that of the Mediatrix (and most other comparable
   fxo devices)
 - don't even think about using spandsp or any other modem-based audio
   through the TDM
 - the latest TDM card revision is rev H. Ensure that is the _only_
   revision shipped to you (many resellers have earlier revisions in
   their stock and most of those are prone to failing every week or
   two requiring a reboot or driver reload)
 - software support for the TDM is almost non-existant from digium and
   from the various * lists (there are only a very small number of
   technical people that understand the drivers, etc, and most won't
   touch them.)

Sipura spa3000:
 - slightly more costly then the TDM card
 - audio quality and echo cancellation about the same as the TDM card
   but below that of the 1204
 - somewhat difficult to configure as there are tons of options that
   are not very well documented, and some options are not fully
   implemented 
 - highly flexible in exactly how you want the fxo and fxs ports to
   be configured and inter-operate with asterisk (most generally 
   limited by your understanding, skills, and the amount of time
   available to experiment)
 - historically, more stable then the TDM card
 - if you think through the US E911 and legal issues associated with
   liability for comm failures, the spa3000 will likely win over 
   any fxo device on the market today for soho environments.
 - much easier to diagnose pstn line problems if you have the skills
   and background to understand what its telling you
 - support sucks and is limited to the reseller (or voxilla.com forum
   users)
 - if you've been around Cisco's internal agendas, there is a high
   probability the sipura devices will improve over time.
 - early boxes seemed to have power issues (high heat generation,
   early failures, wall-wort issues, etc). Later versions seem to
   be fine.
 - fairly good international support (eg, tones, ringing, supervision)
   but probably still needs more work for some country standards.
 - I haven't spent any time testing modem use through the box, so
   no idea how well it might function.

The voip market-space is not well covered in the one-to-five pstn
(fxo) line areana. Above approximately four lines, a channel bank
(with T1 card) is likely to be more cost effective when ongoing
client support and satisfaction is factored into overall costs.

Hope that helps....

Rich

Hello;
i am trying to make a dial plan that can handle any wrong extensions
dialled from the local sip phone for example so that if i dialled the
right extension it rings but if i dialled wrong or existing extension it
redirect him to the Main menu for example?

thanks in advance
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>  i am trying to make a dial plan that can handle any wrong extensions
> dialled from the local sip phone for example so that if i dialled the right
> extension it rings but if i dialled wrong or existing extension it redirect
> him to the Main menu for example?
You might see if you can put the "i" extension to work for you in your
local dialplan.

Hatton

Mahmoud Badran wrote:
> Hello;
> i am trying to make a dial plan that can handle any wrong extensions 
> dialled from the local sip phone for example so that if i dialled the 
> right extension it rings but if i dialled wrong or existing extension 
> it redirect him to the Main menu for example?
>
> thanks in advance 
This seems to be popular at the mo...
Heres one I created earlier... 
http://www.planetwayne.com/forums/viewtopic.php?t=218