[Freeswitch-users] 404 Error on incoming calls

Fri Mar 14 07:26:30 EDT 2008

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:
> 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:
>> 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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>>>
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