[Freeswitch-users] 404 Error on incoming calls
Josip Djuricic
josip.djuricic at primatel.hr
Sat Mar 15 08:45:49 EDT 2008
yeah m8 :)
nice to hear you resolved it
Kurt Marasco wrote:
> We must have been typing at the same time. You are exactly right. Just
> resolved it a bit ago.
> Thank you.
>
>
> Josip Djuricic wrote:
>
>> Kurt if I see it correctly ain't you missing brackets?
>>
>> <condition field="destination_number" expression="^(In-2061234567)$">
>>
>> otherwise $1 has almost no meaning, cause it won't be filled with In-2061234567
>>
>> Josip
>>
>>
>> Kurt Marasco wrote:
>>
>>> Michael, you are absolutely correct that it is matching on the enum
>>> extension in the default profile. When I comment out the enum extension
>>> I get no match.
>>>
>>> So I know that the transfer is being made from public.xml to
>>> default.xml. The question is why am I not getting a match? And is there
>>> any way to see what's contained in the $1 that is being passed from
>>> public,xml to default.xml??? The debug shows $1
>>> (Caller-Destination-Number: [$1]) and not the contents of the variable.
>>> This field shows 1001 when I hard code the final extension.
>>>
>>> I found an error in my extension in default xml relating to where the
>>> condition is closed, but I fixed it and still do not get a match.
>>>
>>> public.xml contains the following:
>>> <!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
>>> <include>
>>> <context name="public">
>>>
>>> <extension name="unloop">
>>> <condition field="${unroll_loops}" expression="^true$"/>
>>> <condition field="${sip_looped_call}" expression="^true$">
>>> <action application="deflect" data="${destination_number}"/>
>>> </condition>
>>> </extension>
>>>
>>> <extension name="public_did2">
>>> <condition field="destination_number" expression="^In-2061234567$">
>>> <action application="transfer" data="$1 XML default"/>
>>> </condition>
>>> </extension>
>>>
>>> </context>
>>> </include>
>>>
>>> The extension public_did2 differs from my working scenario only in that
>>> I hard coded the destination extension 1001 in place of $1.
>>> My extension to match in default.xml is listed prior to the enum
>>> extension that is being matched as follows:
>>> <extension name="In-2061234567">
>>> <condition field="destination_number" expression="^In-2061234567$">
>>> <!-- <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"/>
>>> </condition>
>>> </extension>
>>>
>>> My understanding is that $1 should contain whatever was matched in
>>> public.xml, which is the string "In-2061234567". My condition expression
>>> in default.xml is "^In-2061234567$", so I don't where else to look at
>>> this point. My probable conclusions are that I have a syntax error or
>>> that $1 is passing something other than expected. Extension
>>> "In-2061234567" was not in the directory because it is not a physical
>>> endpoint. I added it to the directory thinking that this might be an
>>> authentication issue, but the result is the same even if "In-2061234567"
>>> is in the directory (No sip phone is registering to "In-2061234567" as
>>> my intent is to transfer/bridge to 1001.
>>>
>>> Kurt
>>>
>>>
>>>
>>> Michael Jerris wrote:
>>>
>>>
>>>> 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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>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
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>>>>>>>> ------------------------------------------------------------------
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>>>>>> ---
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