Well 

freeswitch@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@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@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@129.sip.testy.com>;tag=g0FpDc3651FtN

   To: <sip:tone_detect@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@XX.XX.XX.XX>;party=calling;privacy=off;screen=no

On Tue, Apr 15, 2014 at 12:01 PM, Luis F Urrea <lfurrea@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@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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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