[Freeswitch-users] 404 Error on incoming calls

Fri Mar 14 04:57:59 PDT 2008

Could you please post a FS debug also with sip debug enabled on these event?

Kurt Marasco wrote:
> 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
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org 
>>>>> <mailto:Freeswitch-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>           
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Freeswitch-users mailing list
>>> Freeswitch-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>   
>>>       
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>   

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20080314/7c10955a/attachment-0002.html 