What about modem relay? Should we be concerned when the call goes through the DE410P into asterisk via libss7, and then back out via libpri? Would this degrade the quality of the modem tones to a point the AS5300 will not be able to understand them correctly?<div>
<br></div><div>Any special consideration for this?<br><br><div class="gmail_quote">On Fri, Nov 26, 2010 at 1:44 PM, Cary Fitch <span dir="ltr"><<a href="mailto:sage@usawide.net">sage@usawide.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">SS7 is “simply” another data
channel. As a 56k or 64K channel, it is not a great deal different, as far as
stress goes, than a D channel on a PRI. (Only discussing stress on the
system.) You do have to have two SS7 channels per the SS7 standards, for
reliability. From there on, there should be little to no difference how many
64K channels of audio/modem-tones you can handle. IMO.</span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> </span></font></p>
<p class="MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy">Cary</span></font><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"> Fitch</span></font><font size="2" color="navy" face="Arial"><span style="font-size:10.0pt;font-family:Arial;color:navy"></span></font></p>
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<hr size="2" width="100%" align="center">
</span></font></div>
<p class="MsoNormal"><b><font size="2" color="black" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:bold">From:</span></font></b><font size="2" color="black" face="Tahoma"><span style="font-size:10.0pt;font-family:Tahoma;color:windowtext"> <a href="mailto:asterisk-ss7-bounces@lists.digium.com" target="_blank">asterisk-ss7-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-ss7-bounces@lists.digium.com" target="_blank">asterisk-ss7-bounces@lists.digium.com</a>] <b><span style="font-weight:bold">On Behalf Of </span></b>Marcelo Pacheco<br>
<b><span style="font-weight:bold">Sent:</span></b> Friday, November 26, 2010
12:24 PM<br>
<b><span style="font-weight:bold">To:</span></b> <a href="mailto:asterisk-ss7@lists.digium.com" target="_blank">asterisk-ss7@lists.digium.com</a></span></font></p><font size="2" color="black" face="Tahoma"><div class="im">
<br>
<b><span style="font-weight:bold">Subject:</span></b> Re: [asterisk-ss7]
Incoming calls through SS7 for data modemtransmissions - possible??</div></font><font color="black"><span style="color:windowtext"></span></font><p></p>
</div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt"> </span></font></p>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">I had a quad E1 PCI setup with a PRI to the PSTN and a
PRI to an AS5300, the AS5300 was exactly handling analog modem calls.</span></font></p><font size="3" color="black" face="Times New Roman"><div><div></div><div class="h5"><br>
Works ok, as long as the CPU is always lightly loaded.<br>
Changing one E1 from PRI to SS7 should make no diference whatsoever if you're
using libss7, since libpri uses DAHDI and chan_dahdi for bridging the call.<br>
It was like 6 years ago, so I don't even have the scripts. Just saying it
should work, specially with faster (newer) CPUs.<br>
<br>
José Pablo Méndez Soto wrote: </div></div></font><p></p><div><div></div><div class="h5">
<p class="MsoNormal" style="margin-bottom:12.0pt"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">Thank you Horacio and
Cary.<br>
<br>
We will try receiving SS7, routing via SIP, answering on the AS5300, then
looping back to itself (out PRI, in PRI ports) in order to invoke the modem
termination. This way we may be able to spare the TDM cards in Asterisk and reuse
the E1 ports installed in the gateway.<br>
<br>
Best regards,<br>
<br clear="all">
</span></font><b><font size="2" color="#999999" face="Verdana"><span style="font-size:10.0pt;font-family:Verdana;color:#999999;font-weight:bold">José
Pablo Méndez</span></font></b><b><font size="2" color="#999999"><span style="font-size:10.0pt;color:#999999;font-weight:bold"><br>
<img width="46" height="46"></span></font></b><br>
<br>
</p>
<div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">2010/11/24 Horacio J. Peña <<a href="mailto:horape@compendium.com.ar" target="_blank">horape@compendium.com.ar</a>></span></font></p>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">Hola!<br>
<br>
ZapRAS seems to work only with ISDN calls. "This command is not for use
with<br>
analog lines; it does not provide a modem emulator."<br>
(<a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS</a>)<br>
<br>
You need something doing the modulation. It seems that iaxmodem is your best<br>
bet, and you'll have to make a good bunch of work on it to be able to use as
you<br>
want to.<br>
<br>
If your client has the cisco gateways, I'd suggest you to keep them. They are<br>
very reliable and tested, and with MICA cards they have not a high resale
value,<br>
so you'll probably end with them as paperweights unless you happen to have some<br>
stack of C549 cards to repurpose them.<br>
<br>
Saludos,<br>
H</span></font></p>
<div>
<div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt"><br>
On Wed, Nov 24, 2010 at 07:58:37PM -0600, José Pablo Méndez Soto wrote:<br>
> Hello,<br>
> We are working on implementing a solution for a medium
service<br>
> provider. They were previously using a Cisco AS5300 gateway
with some<br>
> PRI trunks to receive modem calls, then route them out the
Internet.<br>
> The Telco they were buying the trunks from, discovered this<br>
> configuration and restricted them due to legal conventions,
and stated<br>
> that in order to continue doing this, they would have to talk
SS7<br>
> directly.<br>
> We are planning on solving this by placing an Asterisk server
with some<br>
> TE410 cards talking SS7 to Telco, and another 4 ISDN ports
talking to<br>
> the AS5300 for the dial-up to complete after authenticating
against a<br>
> RADIUS server.<br>
> My questions is: can we use only Asterisk to
complete/terminate the<br>
> dial-up connection, removing the AS5300 out of the picture?
We would<br>
> probably need a PPP channel configuration to link the modem
connection<br>
> with the Internet.<br>
> Current topology to be set-up:<br>
> Telco --> SS7 --> TE410P-AsteriskServer --> ISDN
--> AS5300 --><br>
> Internet<br>
> Ideal topology:<br>
> Telco --> SS7 --> TE410P-AsteriskServer --> Internet<br>
> Some posts talk about zapRAS being able to accomplish this,
not quite<br>
> sure though<br>
> Sounds like possible:</span></font></p>
</div>
</div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">> [1]<a href="http://lists.digium.com/pipermail/asterisk-users/2004-January/026956" target="_blank">http://lists.digium.com/pipermail/asterisk-users/2004-January/026956</a><br>
> .html<br>
> [2]<a href="http://lists.digium.com/pipermail/asterisk-users/2009-November/24021" target="_blank">http://lists.digium.com/pipermail/asterisk-users/2009-November/24021</a></span></font></p>
<div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">> 8.html<br>
> Sounds like not possible:</span></font></p>
</div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">> [3]<a href="http://lists.digium.com/pipermail/asterisk-users/2009-November/24020" target="_blank">http://lists.digium.com/pipermail/asterisk-users/2009-November/24020</a></span></font></p>
<div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">> 2.html<br>
> Thanks in advance,<br>
> José Pablo Méndez<br>
></span></font></p>
</div>
<p class="MsoNormal"><font size="3" color="black" face="Times New Roman"><span style="font-size:12.0pt">> References<br>
><br>
> 1. mailto:<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>
> 2. <a href="http://lists.digium.com/pipermail/asterisk-users/2009-November/240218.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/2009-November/240218.html</a><br>
> 3. <a href="http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html" target="_blank">http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html</a><br>
<br>
> --</span></font></p>
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