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<DIV><FONT size=2 face=Arial>Will Do. Thanks very much for your help. I
bought the book:)</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>Dave</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px"
dir=ltr>
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=msc@freeswitch.org href="mailto:msc@freeswitch.org">Michael
Collins</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users Help</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, June 16, 2011 3:22
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] Setting
Up an Sip Trunk without Authentication</DIV>
<DIV><BR></DIV>Okay, I want you to go read up on gather data and dropping it
on pastebin. The skills you hone there will server you well. :)
<DIV><BR></DIV>
<DIV><A
href="http://wiki.freeswitch.org/wiki/Reporting_Bugs">http://wiki.freeswitch.org/wiki/Reporting_Bugs</A></DIV>
<DIV><BR></DIV>
<DIV>Need a debug log and a sip trace. Drop it on pastebin. Hint: use "sofia
global siptrace on" to get all sip traffic logged to the console.</DIV>
<DIV><BR></DIV>
<DIV>Put the pb link in this email thread.</DIV>
<DIV>-MC<BR><BR>
<DIV class=gmail_quote>On Thu, Jun 16, 2011 at 2:18 PM, Dave <SPAN
dir=ltr><<A
href="mailto:dave@clancysystems.com">dave@clancysystems.com</A>></SPAN>
wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2 face=Arial>I did both, and even restarted the service.
same results. Velocity did a call capture on thier end and sent it to me.
They say that FreeSWITCH is requiring proxy authentication and that that is
the issue.</FONT></DIV>
<DIV>SIP Status: 407 Proxy Authentication Required
<P class=MsoNormal> </P></DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px"
dir=ltr>
<DIV class=im>
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV style="FONT: 10pt arial; BACKGROUND: #e4e4e4"><B>From:</B> <A
title=msc@freeswitch.org href="mailto:msc@freeswitch.org"
target=_blank>Michael Collins</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org"
target=_blank>FreeSWITCH Users Help</A> </DIV></DIV>
<DIV>
<DIV></DIV>
<DIV class=h5>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, June 16, 2011 2:46
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users]
Setting Up an Sip Trunk without Authentication</DIV>
<DIV><BR></DIV>Did you reloadxml and restart the sofia profile after the
changes? Also, you need to "reloadacl" after doing a change to
acl.conf.xml.
<DIV><BR></DIV>
<DIV>-MC<BR>
<DIV><BR>
<DIV class=gmail_quote>On Thu, Jun 16, 2011 at 1:27 PM, Dave <SPAN
dir=ltr><<A href="mailto:dave@clancysystems.com"
target=_blank>dave@clancysystems.com</A>></SPAN> wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2 face=Arial>Yes, I am only testing inbound. When
I call from our office phone (not connected to FreeSWITCH, it's on a
trunk from broadvox) I Get a busy signal and at that moment
FreeSWITCH shows...</FONT><FONT size=2 face=Arial></FONT></DIV>
<DIV><FONT size=2 face=Arial>
<DIV><FONT size=2 face=Arial>"SIP auth challenge (INVITE) on sofia
profile 'internal' for [myphone@theirIP] from their IP.</FONT></DIV>
<DIV> </DIV>
<DIV>If I call from my cell phone I get the Operator Message "your call
did not go through" and at that moment the same thing shows in
FreeSWITCH</DIV>
<DIV>
<DIV><FONT size=2 face=Arial>"SIP auth challenge (INVITE) on sofia
profile 'internal' for [myphone@theirIP] from their
IP.</FONT></DIV> </DIV></FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV>
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV style="FONT: 10pt arial; BACKGROUND: #e4e4e4"><B>From:</B> <A
title=msc@freeswitch.org href="mailto:msc@freeswitch.org"
target=_blank>Michael Collins</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org"
target=_blank>FreeSWITCH Users Help</A> </DIV></DIV>
<DIV>
<DIV></DIV>
<DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, June 16, 2011
2:11 PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users]
Setting Up an Sip Trunk without Authentication</DIV>
<DIV><BR></DIV><BR><BR>
<DIV class=gmail_quote>On Thu, Jun 16, 2011 at 1:02 PM, Dave <SPAN
dir=ltr><<A href="mailto:dave@clancysystems.com"
target=_blank>dave@clancysystems.com</A>></SPAN> wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>Under "domains" in acl.conf.xml I added
</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial> <node type="allow"
value="xxx.xxx.xxx.xxx"/></FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV></DIV></BLOCKQUOTE>
<DIV>The correct syntax is:</DIV>
<DIV><SPAN style="FONT-FAMILY: Arial"><node type="allow"
cidr="xxx.xxx.xxx.xxx/32"/></SPAN></DIV>
<DIV> </DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2 face=Arial>x's are the IP address it's just a
single address not a CIDR.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>And I created the file velocity_did.xml
with the following code..</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial><include><BR> <extension
name="velocity_did">
<DIV><BR> <condition field="destination_number"
expression="^(2064001950)$"><BR></DIV>
<action application="set"
data="domain_name=$${domain}"/><BR>
<action application="answer"/>
<DIV><BR> <action application="lua" data="cme_ivr.lua
channel_name"/><BR> <BR> </condition><BR>
</extension><BR></DIV></include><BR></FONT></DIV>
<DIV><FONT size=2 face=Arial>I still just get the "SIP auth
challenge (INVITE) on sofia profile 'internal' for [myphone@theirIP]
from their IP, but nothing more.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV></DIV></BLOCKQUOTE>
<DIV>Wait - can you explain *exactly* what you're doing for testing?
Are you trying to dial out via Velocity and back in to your DID? If so
then you have 2 completely different things you need to set up. The
instructions I gave were only for inbound DID, so use a cell phone or
something to test that.</DIV>
<DIV><BR></DIV>
<DIV>For outbound it seems like they are sending you an auth
challenge, which means they need to give you a username and password
as well as a host/ip. You need to create a gateway, preferably in your
external profile. Just be sure to set the "register" param to false
since they are not expecting you to register with them. </DIV>
<DIV><BR></DIV>
<DIV>-MC<BR><BR></DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2 face=Arial>Do I need to change the line
..</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial><action application="set"
data="domain_name=$${domain}"/><BR></FONT></DIV>
<DIV><FONT size=2 face=Arial>to the IP address? If so what
syntax?</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>They have no Domain Name.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>Thanks again for your
help.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV><FONT color=#888888>
<DIV><FONT size=2 face=Arial>Dave
Goodwin </FONT></DIV></FONT>
<DIV>
<DIV></DIV>
<DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px"
dir=ltr>
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV style="FONT: 10pt arial; BACKGROUND: #e4e4e4"><B>From:</B> <A
title=msc@freeswitch.org href="mailto:msc@freeswitch.org"
target=_blank>Michael Collins</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org"
target=_blank>FreeSWITCH Users Help</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Thursday, June 16, 2011
12:15 PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re:
[Freeswitch-users] Setting Up an Sip Trunk without
Authentication</DIV>
<DIV><FONT size=2 face=Arial></FONT><FONT size=2
face=Arial></FONT><BR></DIV>Dave,
<DIV><FONT size=2 face=Arial></FONT><FONT size=2
face=Arial></FONT><BR></DIV>
<DIV>If you are simply handling incoming calls then you will need
to do two things:</DIV>
<DIV><FONT size=2 face=Arial></FONT><FONT size=2
face=Arial></FONT><BR></DIV>
<DIV>Add Velocity's IP addr to the domains section of
acl.conf.xml</DIV>
<DIV>Add an extension to the public context
(conf/dialplan/public.xml)</DIV>
<DIV><FONT size=2 face=Arial></FONT><FONT size=2
face=Arial></FONT><BR></DIV>
<DIV>Allowing a call into FS via an ACL will send it into the
"public" context; from there you need to transfer it to the
default context or just send it straight to your Lua script.</DIV>
<DIV><FONT size=2 face=Arial></FONT><FONT size=2
face=Arial></FONT><BR></DIV>
<DIV>Also, I don't believe you need to use a "bridge" app based on
the description you gave. Bridge is used to create a new outbound
call leg (B leg) and connect it to the inbound call leg (A leg).
If you just are handling a call with an IVR then there is no B leg
needed, thus no need to bridge.</DIV>
<DIV><FONT size=2 face=Arial></FONT><FONT size=2
face=Arial></FONT><BR></DIV>
<DIV>Welcome to FreeSWITCH, btw! :) I can highly recommend the FS
book (the "bridge" book, ironically) as it discusses some of these
basic concepts that will make your FS experience a whole lot more
pleasant.</DIV>
<DIV><FONT size=2 face=Arial></FONT><FONT size=2
face=Arial></FONT><BR></DIV>
<DIV>-MC<BR><BR>
<DIV class=gmail_quote>On Thu, Jun 16, 2011 at 8:54 AM, Dave <SPAN
dir=ltr><<A href="mailto:dave@clancysystems.com"
target=_blank>dave@clancysystems.com</A>></SPAN> wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2 face=Arial>Hi,</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>I am new to FreeSWITCH and need to
set up FreeSWITCH to receive calls through a Velocity
Networks SIP trunk. They say I should not authenticate
with Username and Password. Rather, I connect directly to the
their IP address.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>I put</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial><extension
name="velocity"><BR>
<condition field="destination_number"
expression="^2064001950$"><BR>
<action application="bridge" data="sofia/external/$ <A
href="mailto:%7Bdestination_number%7D@their_ip_address%22/"
target=_blank>{destination_number}@their_ip_address"/</A>><BR> <action
application="lua" data="cme_ivr.lua
channel_name"/><BR>
</condition><BR></extension></FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>in the default.xml under
Dialplan</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>When I Dial the DID from a Phone
(not one connected to the FreeSWITCH server) I get the "SIP auth
challenge (INVITE) on sofia profile 'internal' for
[myphone@theirIP] from their IP, but nothing more.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>What I hope to do is take the
incoming call from 2064001950 and route it to the Lua IVR script
above.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>I Appreciate any help you
may offer.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2 face=Arial>Dave Goodwin</FONT></DIV>
<DIV> </DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2
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