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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'>I have two MP-108 boxes working fine in
both ways. You have to check these;<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><span style='mso-list:Ignore'>-<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Assign
endpoint phone numbers to FXO ports (Note these numbers you will use in
extensions.conf later)<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><span style='mso-list:Ignore'>-<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Route
calls to asterisk in “Tel to IP Routing” table.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if !supportLists]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><span style='mso-list:Ignore'>-<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>Add endpoint
phone numbers to asterisk’s extensions.conf<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>So whenever a call comes Audiocodes will
dial “endpointphonenumber@asterisk”.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> Mahilal Silva
[mailto:asteriskcrazy@gmail.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, June 19, 2006 10:58
PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Asterisk-Users] Help
with Audicodes MP-104</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Not quite sure.
Audiocodes gives a dialtone when the number is called from PSTN. After few
seconds I see the SIP invite to the Asterisk box. Asterisk responds with SIP
404 .<br>
<br>
Thanks,<br>
Lal<o:p></o:p></span></font></p>
<div>
<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 6/12/06, <b><span style='font-weight:bold'>Erick
Perez</span></b> <<a href="mailto:eaperezh@gmail.com">eaperezh@gmail.com</a>>
wrote:</span></font></span><o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>So is the problem with your audiocodes or with the asterisk system?<br>
if it is with the asterisk, what kind of calls are you trying route to<br>
your box? SIP/IAX/other?<br>
<br>
On 6/12/06, Mahilal Silva <<a href="mailto:asteriskcrazy@gmail.com">
asteriskcrazy@gmail.com</a>> wrote:<br>
><br>
> Hi All<br>
> I have been able to get MP 104 FXO to make outbound calls with my asterisk<br>
> box and polycom IP 500 phone.<br>
> However I cannot get the incoming calls to hit the asterisk box. <br>
> Any help will be appreciated.<br>
><br>
> Thanks,<br>
> Lal<br>
> _______________________________________________<br>
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><br>
><br>
<br>
<br>
--<br>
------------------------------------------------------------<br>
Erick Perez<br>
<st1:country-region w:st="on"><st1:place w:st="on">Panama</st1:place></st1:country-region>
Sistemas<br>
Integradores de Telefonia IP y Soluciones Para Centros de Datos<br>
<st1:country-region w:st="on">Panama</st1:country-region>, Republica de Panama <br>
Cel <st1:country-region w:st="on"><st1:place w:st="on">Panama</st1:place></st1:country-region>.
+(507) 6694-4780<br>
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