[Freeswitch-users] 404 Error on incoming calls
Kurt Marasco
kmarasco at faithwork.org
Sat Mar 15 01:31:15 PDT 2008
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
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>>>>>>
>>>>>>
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