[Freeswitch-dev] Freeswitch signaling issue when bridge call from FS to asterisk server.

[Peter Olsson]

If it's not a development related question it should go to the users list.
A question regarding help to build your own module in C would be a
development related question, this is not.

Please post to users list instead, you will get lots of more attention in
there.

/Peter


2013/8/19 ThanhTruong <ngoc217thanh at yahoo.com>

> Hi all,
>
> That is not a basic problem.
>
> Because FS and Asterisk server not in NAT. I can call direct when register
> all softphone to asterisk server.
>
> Thanks,
>
> ThanhTruong
>
>   ------------------------------
>  *From:* Michael Jerris <mike at jerris.com>
> *To:* freeswitch-dev at lists.freeswitch.org
> *Sent:* Monday, August 19, 2013 8:36 PM
> *Subject:* Re: [Freeswitch-dev] Freeswitch signaling issue when bridge
> call from FS to asterisk server.
>
> seems like a freeswitch-users question to me.
>
> On Aug 19, 2013, at 6:53 AM, ThanhTruong <ngoc217thanh at yahoo.com> wrote:
>
> Hi all,
>
> I have a small issue with my testing system with asterisk and FS server.
>
> I have 2 servers. One is Fs and one is asterisk server. I open incoming
> call from asterisk server(for all call coming).
>
> In Fs server, I have routed call in public.xml like:
>
> <extension name="ROUTE">
>         <condition field="destination_number" expression="(\d+)$"
> require-nested="true">
>             <condition field="network_addr" expression="^X\.X\.X\.X$"
>  break="never">
>                 <action application="set" data="hangup_after_bridge=true"/>
>                 <action application="export" data="sip_h_X-ROUTE=LOOKUP"/>
>                 <action application="bridge"
> data="user/${destination_number}@${domain_name}"/>
>                 <anti-action application="set"
> data="hangup_after_bridge=true"/>
>                 <anti-action application="export"
> data="sip_h_X-ROUTE=LOOKUP"/>
>                 <anti-action application="bridge"
> data="sofia/internal/${destination_number}@X.X.X.X"/>
>             </condition>
>         </condition>
>     </extension>
>
> I have 2 client :
>
> + Xlite registered in Fs server
> + Eyebeam registered in asterisk server.
>
> Then I have tried some several case and get some results as:
>
> + when I make call from xlite to eyebeam , call connected and:
>
>    - When i hangup from xlite, eyebeam keep connected and not hangup
>    - when I hangup from eyebeam, xlite keep connected and not hangup too.
>
> +Then, I have tried to make call from eyebeam to xlite, call connected and:
>
>     - I hangup call in xlite, eyebeam hangup.
>     - I hangup call in eyebeam, xlite hangup.
>
>  -----------------------> only can hangup in this case.
>
> + Then I have tried another case:
> I have call from eyebeam to xlite and vise versace : call auto hangup
> after 30 seconds.
> I have thought that is NAT issue, But If I register direct xlite and
> eyebeam to asterisk server, call will ok, and there is not any issue there.
> So, Hope some one can suggest some thing to fix it.
>
> Thanks,
>
> ThanhTruong
> _________________________________________________________________________
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>
> 
> 
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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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-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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