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Could you please post a FS debug also with sip debug enabled on these
event?<br>
<br>
<br>
Kurt Marasco wrote:
<blockquote cite="mid:47DA60E6.7060105@faithwork.org" type="cite">
<pre wrap="">Thanks,
That's what I tried, but I ended up I hardcoding the extension. When I
left it as $1, the console showed it being passed literally as $1 (But
maybe it was actually passing the contents of the variable). Since I
don't have a registered endpoint that matched my incoming DID, I was
trying to have the default dial plan handle the passed DID with no luck.
It seemed to only want to transfer or bridge to a registered endpoint
and not into the dial plan.
Perhaps my syntax in the default dial plan was wrong. I tried this in
the public.xml:
<extension name="public_did2">
<condition field="destination_number" expression="^(In-2061234567)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
And this in the default dial plan:
<extension name="In-2061234567">
<condition field="destination_number" expression="^In-2061234567$"/>
<action application="ringback" />
<action application="set" data="call_timeout=20"/>
<action application="bridge" data="sofia/default/1001%$${domain}"/>
<action application="javascript"
data="/usr/local/freeswitch/scripts/answermachine.js"/>
</extension>
The above fails, but below worked by itself in public.xml:
<extension name="public_did">
<condition field="destination_number" expression="^In-2061234567$">
<action application="transfer" data="1001 XML default"/>
</condition>
</extension>
Thanks,
Kurt
Josip Djuricic wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi there,
if I'm not mistaking (if I am Brian or someone else will tell), you
can do it from the public.xml
Example:
<extension name="name_of_incoming_extension">
<condition field="destination_number"
expression="^(incoming_extension_number_match)$">
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
If I'm not mistaking with transfer to XML default you do exactly what
you wanna do.
Josip
Kurt Marasco wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Thanks Brian and Josip for your responses,
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.
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.
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.
Brian West wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Kurt,
First off let me fill in a few blanks here.
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)
Here is what you do. Have your IPKALL did hit your IP on port 5080
instead.. aka the outbound profile.
Then open up dialplan/public.xml and install an extension that can
route to a registered endpoing. their is a 5551212 example in there.
/b
On Mar 11, 2008, at 5:02 AM, Kurt Marasco wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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.
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.
Here is the console output from FS when I dial my IpKall DID from
my land line.
</pre>
<blockquote type="cite">
<pre wrap="">nta: received INVITE <a class="moz-txt-link-freetext" href="sip:In-2061234567@mydomain.com:5070">sip:In-2061234567@mydomain.com:5070</a> SIP/2.0
(CSeq 102)
nta: INVITE (102) going to a default leg
nua(0x8117508): adding session usage
nta: sent 100 Trying for INVITE (102)
nua(0x8117508): call state changed: init -> received, received offer
2008-03-11 02:25:31 [NOTICE] switch_channel.c:522
switch_channel_set_name() New Chan
<a 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]
2008-03-11 02:25:32 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
Processing PORTLAND OR->In-2061234567!
*2008-03-11 02:25:32 [INFO] switch_core_state_machine.c:112
switch_core_standard_on_ring() No Route, Aborting*
2008-03-11 02:25:32 [NOTICE] switch_core_state_machine.c:113
switch_core_standard_on_ring() Hangup
<a 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]
nta: sent 404 Not Found for INVITE (102)
nua(0x8117508): removing session usage
nua(0x8117508): call state changed: init -> terminated
nta: received ACK <a class="moz-txt-link-freetext" href="sip:In-2061234567@mydomain.com:5070">sip:In-2061234567@mydomain.com:5070</a> SIP/2.0
(CSeq 102)
nta: ACK (102) is going to INVITE (102)
2008-03-11 02:25:32 [NOTICE] switch_core_session.c:717
switch_core_session_thread() Session 1
(<a 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
2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719
switch_core_session_thread() Close Channel
<a 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]
</pre>
</blockquote>
<pre wrap="">Any thoughts on what I'm doing wrong would be appreciated.
Kurt
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</blockquote>
</blockquote>
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<pre wrap=""><!---->
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