[Freeswitch-users] 404 Error on incoming calls

Michael Jerris mike at jerris.com
Sat Mar 15 00:28:40 EDT 2008 

This is a different extension that is transfering to the enum dialplan  
not XML like the extension you pasted.  Chek your conditions to see  
why its matching the wrong ext.

Mike

On Mar 14, 2008, at 6:31 PM, Kurt Marasco <kmarasco at faithwork.org>  
wrote:

> Here's the FS debug from the console:
>
> nta: received INVITE sip:In-2061234567 at mydomain.com:5080 SIP/2.0  
> (CSeq 102)
> nta: INVITE (102) going to a default leg
> nua(0x81c22c0): adding session usage
> nta: sent 100 Trying for INVITE (102)
> nua(0x81c22c0): call state changed: init -> received, received offer
> 2008-03-14 18:18:40 [NOTICE] switch_channel.c:522
> switch_channel_set_name() New Chan
> sofia/outbound/5031234567 at 67.55.341.56:5060
> [b9608e95-8a80-49ba-b3c5-883432c4bcb2]
> 2008-03-14 18:18:40 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
> Processing PORTLAND OR->In-2061234567!
> 2008-03-14 18:18:40 [NOTICE] switch_ivr.c:924
> switch_ivr_session_transfer() Transfer
> sofia/outbound/5031234567 at 67.55.341.56:5060 to XML[$1 at default]
> 2008-03-14 18:18:40 [INFO] mod_dialplan_xml.c:222 dialplan_hunt()
> Processing PORTLAND OR->$1!
> 2008-03-14 18:18:40 [INFO] mod_dptools.c:601 info_function()  
> CHANNEL_DATA:
> Channel-State: [CS_EXECUTE]
> Channel-State-Number: [4]
> Channel-Name: [sofia/outbound/5031234567 at 67.55.341.56:5060]
> Unique-ID: [b9608e95-8a80-49ba-b3c5-883432c4bcb2]
> Call-Direction: [inbound]
> Answer-State: [ringing]
> Channel-Read-Codec-Name: [PCMU]
> Channel-Read-Codec-Rate: [8000]
> Channel-Write-Codec-Name: [PCMU]
> Channel-Write-Codec-Rate: [8000]
> Caller-Username: [5031234567]
> Caller-Dialplan: [XML]
> Caller-Caller-ID-Name: [PORTLAND OR]
> Caller-Caller-ID-Number: [5031234567]
> Caller-Network-Addr: [67.55.341.56]
> Caller-Destination-Number: [$1]
> Caller-Unique-ID: [b9608e95-8a80-49ba-b3c5-883432c4bcb2]
> Caller-Source: [mod_sofia]
> Caller-Context: [default]
> Caller-RDNIS: [In-2061234567]
> Caller-Channel-Name: [sofia/outbound/5031234567 at 67.55.341.56:5060]
> Caller-Channel-Created-Time: [1205543920601430]
> Caller-Channel-Answered-Time: [0]
> Caller-Channel-Hangup-Time: [0]
> Caller-Channel-Transfer-Time: [0]
> Caller-Screen-Bit: [yes]
> Caller-Privacy-Hide-Name: [no]
> Caller-Privacy-Hide-Number: [no]
> variable_sip_from_user: [5031234567]
> variable_sip_from_port: [5060]
> variable_sip_from_uri: [5031234567 at 67.55.341.56:5060]
> variable_sip_from_host: [67.55.341.56]
> variable_sip_from_user_stripped: [5031234567]
> variable_sip_from_tag: [as6a5e4b5c]
> variable_sofia_profile_name: [outbound]
> variable_sofia_profile_domain_name: [outbound]
> variable_sip_req_user: [In-2061234567]
> variable_sip_req_port: [5080]
> variable_sip_req_uri: [In-2061234567 at mydomain.com:5080]
> variable_sip_req_host: [mydomain.com]
> variable_sip_to_user: [In-2061234567]
> variable_sip_to_port: [5080]
> variable_sip_to_uri: [In-2061234567 at mydomain.com:5080]
> variable_sip_to_host: [mydomain.com]
> variable_sip_contact_user: [5031234567]
> variable_sip_contact_port: [5060]
> variable_sip_contact_uri: [5031234567 at 67.55.341.56:5060]
> variable_sip_contact_host: [67.55.341.56]
> variable_channel_name: [sofia/outbound/5031234567 at 67.55.341.56:5060]
> variable_sip_call_id: [637c71185629eca135630a1d3c215d08 at 67.55.341.56]
> variable_sip_user_agent: [Asterisk PBX]
> variable_sip_via_host: [67.55.341.56]
> variable_sip_via_port: [5060]
> variable_sip_via_rport: [5060]
> variable_max_forwards: [70]
> variable_switch_r_sdp: [v=0
> o=root 25187 25187 IN IP4 67.55.341.56
> s=session
> c=IN IP4 67.55.341.56
> t=0 0
> m=audio 12740 RTP/AVP 0 8 3 18 97 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> ]
> variable_remote_media_ip: [67.55.341.56]
> variable_remote_media_port: [12740]
> variable_read_codec: [PCMU]
> variable_read_rate: [8000]
> variable_write_codec: [PCMU]
> variable_write_rate: [8000]
> variable_endpoint_disposition: [RECEIVED]
> variable_use_profile: [default]
> variable_numbering_plan: [US]
> variable_default_gateway: [192.168.2.1]
> variable_default_area_code: [509]
> variable_user_name: [default]
> variable_domain_name: [192.168.2.102]
>
>
> 2008-03-14 18:18:41 [NOTICE] switch_ivr.c:924
> switch_ivr_session_transfer() Transfer
> sofia/outbound/5031234567 at 67.55.341.56:5060 to enum[$1 at default]
> 2008-03-14 18:18:41 [INFO] switch_core_state_machine.c:112
> switch_core_standard_on_ring() No Route, Aborting
> 2008-03-14 18:18:41 [NOTICE] switch_core_state_machine.c:113
> switch_core_standard_on_ring() Hangup
> sofia/outbound/5031234567 at 67.55.341.56:5060 [CS_RING]  
> [NO_ROUTE_DESTINATION]
> nta: sent 404 Not Found for INVITE (102)
> nua(0x81c22c0): removing session usage
> nua(0x81c22c0): call state changed: init -> terminated
> 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:717
> switch_core_session_thread() Session 2
> (sofia/outbound/5031234567 at 67.55.341.56:5060) Ended
> 2008-03-14 18:18:41 [NOTICE] switch_core_session.c:719
> switch_core_session_thread() Close Channel
> sofia/outbound/5031234567 at 67.55.341.56:5060 [CS_HANGUP]
>
> Thanks
>
> Josip Djuricic wrote:
>> 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
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>>>>>>> http://www.freeswitch.org
>>>>>>>
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>>>>>
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