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<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=022471713-22082005>Chris,</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=022471713-22082005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>Thank
you for your answer. By the way, my * server won't be a PSTN gateway. The SER is
connected to another SIP gateway provided by our Telco. Would you be so kind to
give me some more details on this:</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=022471713-22082005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>- Say
I have * server A with extensions 100, 101, 102, and * server B also with
extensions starting from 100, .... (Server A is for another company as
Server B)</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=022471713-22082005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>How
can I make these phones register themselves with SER. Or do I need a unique
username for each extension (say username: companyA100, ... and username
companyB100)</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>How
will I forward all the calls to a * server. So how do I forward the calls from
companyA100 to * server A etc...</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>Do the
* servers need to be registered at the SER server?</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=022471713-22082005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>Can
you please provide me with some example configs (both on the SER as on the *
side? I cannot find to many examples on this one (besides the 'Asterisk at
large'-WIKI)...</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=022471713-22082005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>Thank
you very much,</SPAN></FONT></DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN
class=022471713-22082005></SPAN></FONT> </DIV>
<DIV><FONT face=Arial color=#0000ff size=2><SPAN class=022471713-22082005>Ronald
Voermans</SPAN></FONT></DIV>
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<FONT face=Tahoma size=2><B>Van:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>Namens </B>Chris
HARIGA<BR><B>Verzonden:</B> maandag 22 augustus 2005 15:10<BR><B>Aan:</B>
'Asterisk Users Mailing List - Non-Commercial Discussion'<BR><B>Onderwerp:</B>
RE: [Asterisk-Users] Nat + Asterisk + Ser (Far end Nat
Traversal)<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=ltr align=left><SPAN class=692470513-22082005><FONT face=Arial
color=#0000ff size=2>Hi,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=692470513-22082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=692470513-22082005><FONT face=Arial
color=#0000ff size=2>all your phones need to be register with SER. The
asterisk will be just PSTN gateway, voicemail server or something else (I prefer
to forward all the calls from ser to asterisk because it's easy to manage the
dialplan). I have the same configuration, I balance the traffic with SER
and I use realtime with asterisk servers.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=692470513-22082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=692470513-22082005><FONT face=Arial
color=#0000ff size=2>Best regards,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=692470513-22082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=692470513-22082005><FONT face=Arial
color=#0000ff size=2>Chris HARIGA</FONT></SPAN></DIV>
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color=#0000ff size=2></FONT></SPAN><FONT face=Arial color=#0000ff
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