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Hi there,<br>
<br>
if I'm not mistaking (if I am Brian or someone else will tell), you can
do it from the public.xml<br>
<br>
Example:<br>
<extension name="name_of_incoming_extension"><br>
<condition field="destination_number"
expression="^(incoming_extension_number_match)$"><br>
<action application="transfer" data="$1 XML default"/><br>
</condition><br>
</extension><br>
<br>
If I'm not mistaking with transfer to XML default you do exactly what
you wanna do.<br>
<br>
Josip<br>
<br>
Kurt Marasco wrote:
<blockquote cite="mid:47DA56A4.3070903@faithwork.org" type="cite">
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<title></title>
Thanks Brian and Josip for your responses,<br>
<br>
Brian's suggestion did the trick for me. I can both transfer and bridge
the call to a registered extension in the default dial plan.<br>
<br>
Not sure if If it makes sense to do this, but is there a way to pass
the call into the default dial plan and have the default dial plan
process the sip invite. I'm able to send the incoming did to a
registered endpoint from (in the directory) but can't pass it through
to the default and match on the original incoming did. <br>
<br>
I'm still confused about what the nat profile does, because I'm behind
nat and am not using the nat profile, yet freeswitch seems to be
working.<br>
<br>
<br>
Brian West wrote:
<blockquote
cite="mid:%3CA1E72750-78A8-4830-B241-22E6C088589A@mac.com%3E"
type="cite">Kurt,
<div><span class="Apple-tab-span" style="white-space: pre;"> </span>First
off let me fill in a few blanks here.</div>
<div><br class="webkit-block-placeholder">
</div>
<div>Correct me if i'm wrong this looks like an inbound invite to
port 5070 right? If so then you're not using the default config as it
was designed. (I did the bulk of the config)</div>
<div><br class="webkit-block-placeholder">
</div>
<div>Here is what you do. Have your IPKALL did hit your IP on port
5080 instead.. aka the outbound profile. </div>
<div><br class="webkit-block-placeholder">
</div>
<div>Then open up dialplan/public.xml and install an extension that
can route to a registered endpoing. their is a 5551212 example in
there.</div>
<div><br class="webkit-block-placeholder">
</div>
<div>/b</div>
<div><br class="webkit-block-placeholder">
</div>
<div><br>
<div>
<div>On Mar 11, 2008, at 5:02 AM, Kurt Marasco wrote:</div>
<br class="Apple-interchange-newline">
<blockquote type="cite">
<div bgcolor="#ffffff" text="#000000"> Hi I am testing FS and am
currently working with the xml dialplan. I have FS behind a NAT router
and have 2 soft phones functioning on another PC behind the router. I
currently have working conversations when dialing between the
extensions set up on each phone. <br>
<br>
I am now trying to call one of the softphones via an IpKall DID. I have
no problem making this work if I use wikipbx, but can't make it work
using the xml dialplan, so clearly FS is working and my configuration
is the issue. I am currently sending the ipkall sip invite to port
5070, but have tried 5060 as well.<br>
<br>
Here is the console output from FS when I dial my IpKall DID from my
land line.<br>
<br>
<blockquote type="cite">nta: received INVITE <a
moz-do-not-send="true" class="moz-txt-link-freetext"
href="sip:In-2061234567@mydomain.com:5070">sip:In-2061234567@mydomain.com:5070</a>
SIP/2.0 (CSeq 102)<br>
nta: INVITE (102) going to a default leg<br>
nua(0x8117508): adding session usage<br>
nta: sent 100 Trying for INVITE (102)<br>
nua(0x8117508): call state changed: init -> received, received offer<br>
2008-03-11 02:25:31 [NOTICE] switch_channel.c:522
switch_channel_set_name() New Chan <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a>
[53bb0a56-f059-483e-9e08-d583a9566255]<br>
2008-03-11 02:25:32 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Processing PORTLAND OR->In-2061234567!<br>
<b>2008-03-11 02:25:32 [INFO] switch_core_state_machine.c:112
switch_core_standard_on_ring() No Route, Aborting</b><br>
2008-03-11 02:25:32 [NOTICE] switch_core_state_machine.c:113
switch_core_standard_on_ring() Hangup <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a>
[CS_RING] [NO_ROUTE_DESTINATION]<br>
nta: sent 404 Not Found for INVITE (102)<br>
nua(0x8117508): removing session usage<br>
nua(0x8117508): call state changed: init -> terminated<br>
nta: received ACK <a moz-do-not-send="true"
class="moz-txt-link-freetext"
href="sip:In-2061234567@mydomain.com:5070">sip:In-2061234567@mydomain.com:5070</a>
SIP/2.0 (CSeq 102)<br>
nta: ACK (102) is going to INVITE (102)<br>
2008-03-11 02:25:32 [NOTICE] switch_core_session.c:717
switch_core_session_thread() Session 1 (<a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a>)
Ended<br>
2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719
switch_core_session_thread() Close Channel <a moz-do-not-send="true"
class="moz-txt-link-abbreviated"
href="mailto:sofia/nat/5035557777@69.64.180.77:5060">sofia/nat/5035557777@69.64.180.77:5060</a>
[CS_HANGUP]<br>
</blockquote>
Any thoughts on what I'm doing wrong would be appreciated.<br>
<br>
Kurt<br>
</div>
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</blockquote>
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