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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'>
asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b><span style='font-weight:bold'>On Behalf Of </span></b>Walid Azab<br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, July 05, 2005 4:23
AM<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> [Asterisk-Users] Asterisk
on Linksys WRT54G</span></font></p>
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12.0pt'> </span></font></p>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hi all,</span></font></p>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Any one tried installing Asterisk on Linksys WRT54G? We have
but facing problems with SIP to SIP calls. The phones ring and calls are
established but we cannot hear any voice at all. I tried allow=all in the
general section but did not work. So I forced ulaw. Can any one please check it
out and let me know what is wrong?</span></font></p>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Here are the conf files:</span></font></p>
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12.0pt'> </span></font></p>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><strong><b><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>Asterisk Version:</span></font></b></strong><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'> Asterisk
CVS-HEAD-01/17/05-00:35:58 built by <a href="mailto:root@splurge">root@splurge</a>
on a i686 running Linux</span></font></p>
</div>
<div>
<p class=MsoNormal><strong><b><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>==>SIP.CONF</span></font></b></strong></p>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>[general]</span></font></p>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>port =
5060 ; Port to bind
to (SIP is 5060)<br>
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on
machine)<br>
disallow=all
; Allow all codecs<br>
allow=ulaw<br>
context = bogon-calls ; Send SIP callers that we don't know about here</span></font></p>
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12.0pt'> </span></font></p>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>[2000]</span></font></p>
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12.0pt'> </span></font></p>
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<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>type=friend
; This device takes and makes calls<br>
username=2000 ; Username on
device<br>
secret=1234 ;
Password for device<br>
host=dynamic ; This host
is not on the same IP addr every time<br>
context=from-sip ; Inbound calls from this host
go here<br>
mailbox=100 ;
Activate the message waiting light if this<br>
; voicemailbox has messages in it</span></font></p>
</div>
<div>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>[2001]
; Duplicate of 2000, except with different auth data</span></font></p>
</div>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>type=friend<br>
username=2001<br>
secret=1234<br>
host=dynamic<br>
context=from-sip<br>
mailbox=101</span></font></p>
</div>
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12.0pt'> </span></font></p>
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<div>
<p class=MsoNormal><strong><b><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>==>Extensions.conf</span></font></b></strong></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>[general]</span></font></p>
</div>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>static=yes <br>
writeprotect=yes </span></font></p>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>[bogon-calls]</span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>exten => _.,1,Congestion</span></font></p>
</div>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>[from-sip]</span></font></p>
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<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>exten => 2000,1,Dial(SIP/2000,20)</span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>exten => 2000,2,Voicemail(u2000)</span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>exten => 2000,102,Voicemail(b2000)<br>
exten => 2000,103,Hangup</span></font></p>
</div>
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12.0pt'> </span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>exten => 2001,1,Dial(SIP/2001,20)<br>
exten => 2001,2,Voicemail(u2001)<br>
exten => 2001,102,Voicemail(b2001)<br>
exten => 2001,103,Hangup</span></font></p>
</div>
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12.0pt'> </span></font></p>
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<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>exten => 2999,1,VoicemailMain(${CALLERIDNUM})</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'> </span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>How are the routers connected to the IP
network? Any nat before them on either end?</span></font></p>
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