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<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2>you can do that, in my experiencie, using oh323 I could not
handle more than 30 active calls, doing g729 passthru...</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2>I dont know how to do IP limitation.... for restrict ip
access use iptables....</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2>I did basic dialpeers like this:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2>exten -> 1305.,1,dial(OH323/xxxx)</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2>exten -> 1305.,2,congestion</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=783460019-19072004><FONT face=Arial
color=#0000ff size=2>I am right?</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=es dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>De:</B> asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] <B>En nombre de </B>Alejandro
Sosa<BR><B>Enviado el:</B> Lunes, 19 de Julio de 2004 03:25 p.m.<BR><B>Para:</B>
asterisk-users@lists.digium.com<BR><B>Asunto:</B> [Asterisk-Users] PSTN gateway
implementation?<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=Section1>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Hello,</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">I need help in creating a simple PSTN Gateway. This is
the scenario:</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">-I have one client sending me VoIP traffic (they don’t
have asterisk, so IAX is out of the picture for me) and I need to validate that
traffic (only accept calls coming from his IP). After that I would terminate the
calls to the PSTN network and keep logs for billing purposes.</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">-I have a TE405P board and only one T1 worth of phone
lines (24) connected to it using an Adtran TA750 channel bank.</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">Is Asterisk capable of handling multiple incoming VoIP
calls arriving from the same source (IP) or do I need to get something else to
take the incoming traffic and pass it on to Asterisk? (I’ve read about using SER
as a SIP proxy, but it’s not clear to me wheather I need it or not). Can I use
the OpenH.323 module to take care of the incoming VoIP
traffic?</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">I am totally new to all this. Any help will be really
appreciated.</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">Please give me your input on which is the best
implementation of a VoIP->PSTN gateway and the some sample configuration
files for the plattforms involved (Asterisk, SER, OpenH323,
etc.)</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt">Thanks in advance.</SPAN></FONT></P>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">Alejandro Sosa.</SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"></SPAN></FONT> </P></DIV></BODY></HTML>