[Freeswitch-users] Inbound call through Registered gateway

[Luis F Urrea]

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.
>>>>
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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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
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