[Freeswitch-users] 404 Error on incoming calls
Kurt Marasco
kmarasco at faithwork.org
Fri Mar 14 21:31:59 EDT 2008
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>
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>>>>>>
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