[Freeswitch-users] 404 Error on incoming calls

[Kurt Marasco]

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:
> 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:
>
>> 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.
>>
>>> nta: received INVITE sip:In-2061234567 at mydomain.com:5070 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 
>>> sofia/nat/5035557777 at 69.64.180.77:5060 
>>> [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 
>>> sofia/nat/5035557777 at 69.64.180.77:5060 [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 sip:In-2061234567 at mydomain.com:5070 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 
>>> (sofia/nat/5035557777 at 69.64.180.77:5060) Ended
>>> 2008-03-11 02:25:32 [NOTICE] switch_core_session.c:719 
>>> switch_core_session_thread() Close Channel 
>>> sofia/nat/5035557777 at 69.64.180.77:5060 [CS_HANGUP]
>> Any thoughts on what I'm doing wrong would be appreciated.
>>
>> Kurt
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