[Freeswitch-users] Inbound call through Registered gateway

[Steven Ayre]

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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