[Freeswitch-users] Inbound call through Registered gateway
Steven Ayre
steveayre at gmail.com
Wed Apr 16 03:50:35 MSD 2014
>
> 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
>>>
>>>
>>>
>>>
>>> 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
>> 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
> 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20140416/82a6ed92/attachment.html
Join us at ClueCon 2013 Aug 6-8, 2013
More information about the FreeSWITCH-users
mailing list