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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 still do not understand how to do this.
I have reviewed the documentation but for example don’t see a place to
input my password in sip.conf. I have the following settings:<br>
<br>
sip proxy</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Sip domain</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Userid</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Password</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'>I can configure my service with the estara
soft phone by inputting these values and make and receive calls.</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'>I can also make and receive multiple calls
at the same time, would this somehow be supported by Asterisk (IE: I have “unlimited”
lines, I am just billed by the minute for each call that takes place, will
asterisk be able to handle more than one call, bandwidth permitting?)</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 style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b>
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Andrew Joakimsen<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, August 08, 2003 2:14
AM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] SIP
Lines</span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'> </span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Instead of using a PCI card is it
possible to use an outside SIP service for “CO” lines?</span></font></p>
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