[Freeswitch-users] Inbound call through Registered gateway

Wed Apr 16 04:02:26 MSD 2014

Right what I was trying to say is that I am running this through the
foreground console, and the output I pasted is exactly what is happening
with that inbound call.

On Tue, Apr 15, 2014 at 4:50 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Enable the debug-level log ("/log 9" in fs_cli)
>
>
> The command you ran affects mod_console which is the foreground console.
> fs_cli connects to the remote console via mod_event_socket, for it you set
> the logging level within your fs_cli client with its local /log command
> since different ESL clients can be using different logging levels.
>
> "/log 9" will show all logging messages
>
>
> On 15 April 2014 23:50, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> Well
>>
>> freeswitch at sipzag> console loglevel
>>
>> +OK console log level set to DEBUG
>>
>> However I only see output from the sofia siptrace on not regular
>> freeswitch logs.
>>
>> Does that even make sense?
>>
>>
>> On Tue, Apr 15, 2014 at 2:44 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Enable the debug-level log ("/log 9" in fs_cli) and show us everything
>>> that happens when the INVITE is received.
>>>
>>>
>>> On 15 April 2014 22:34, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>
>>>> The INVITE is just dropped with a 408, I don't see any dialplan logging
>>>> output.
>>>>
>>>> Your help is appreciated on this.
>>>>
>>>>
>>>> SIP/2.0 480 Temporarily Unavailable
>>>>    Via: SIP/2.0/UDP XX.XX.XX.XX:7000;branch=z9hG4bK0b7b.17642242.0
>>>>    Via: SIP/2.0/UDP
>>>> XX.XX.XX.XX:11000;received=10.159.12.163;rport=11000;branch=z9hG4bKg1HZ4F53ZUD8a
>>>>    From: "Sangoma Technologies" <sip:6502626901 at 129.sip.testy.com
>>>> >;tag=g0FpDc3651FtN
>>>>    To: <sip:tone_detect at XX.XX.XX.XX:61179>;tag=UeDtcKU7ZaU3S
>>>>    Call-ID: 05d75f66-c4e5-11e3-9d0d-9b9b93df5131
>>>>    CSeq: 58462619 INVITE
>>>>    User-Agent:
>>>> FreeSWITCH-mod_sofia/1.5.12b+git~20140410T213613Z~f1d7721710~64bit
>>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY
>>>>    Supported: timer, path, replaces
>>>>    Allow-Events: talk, hold, conference, refer
>>>>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>>>    Content-Length: 0
>>>>    Remote-Party-ID: "tone_detect" <sip:tone_detect at XX.XX.XX.XX
>>>> >;party=calling;privacy=off;screen=no
>>>>
>>>>
>>>> On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea at gmail.com>wrote:
>>>>
>>>>> The point is that the gateway registers using a random port as the
>>>>> source port 52762 and then when the INVITE comes how would that be
>>>>> associated with the SIP profile?
>>>>>
>>>>>
>>>>> On Mon, Apr 14, 2014 at 5:03 PM, Luis F Urrea <lfurrea at gmail.com>wrote:
>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>> I am trying to receive a call through a gateway that is suing
>>>>>> registration on external profile and I don't clearly understand how to
>>>>>> configure the extension to receive the call.
>>>>>>
>>>>>> I appears from the example.xml gateway configuration that the an
>>>>>> extension name with the username used for gateway registration can be used
>>>>>> to get the dialplan to accept the call.
>>>>>>
>>>>>>  I assume  this  needs to be configured on the public context,
>>>>>> however I see the INVITE coming in, but the dialplan never kicks in.
>>>>>>
>>>>>> Do I have to receive the INVITE at port 5080? If that is the case,
>>>>>> registration doesn't really make sense.
>>>>>>
>>>>>> public.xml
>>>>>>
>>>>>> <extension name="tone_detect">
>>>>>>       <condition field="destination_number"
>>>>>> expression="^tone_detect$">
>>>>>>         <action application="pre_answer"/>
>>>>>>         <action application="sleep" data="20000"/>
>>>>>>         <action application="answer"/>
>>>>>>         <action application="sleep" data="1000"/>
>>>>>>         <action application="playback"
>>>>>> data="voicemail/vm-goodbye.wav"/>
>>>>>>         <action application="hangup"/>
>>>>>>       </condition>
>>>>>>     </extension>
>>>>>>
>>>>>>
>>>>>> Gateway config
>>>>>>
>>>>>> <include>
>>>>>>   <gateway name="tone_detect">
>>>>>>   <param name="username" value="tone_detect"/>
>>>>>>   <...snip...>
>>>>>> </include>
>>>>>>
>>>>>> Thanks in advance for your help.
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>>
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>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> 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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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