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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 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 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 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 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 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 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>
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