[Freeswitch-users] Freeswitch as a SIP client

David Villasmil david.villasmil.work at gmail.com
Thu Sep 29 13:31:18 MSD 2016


this is what my gateway look like:

  <gateways>
        <gateway name="whatever">
                <param name="username" value="1001"/>
                <param name="realm" value="THE REALM AGAINST UPON WHICH THE
HASH WILL BE BUILT"/>
                <param name="password" value="password"/>
                <param name="extension" value="auto_to_user"/>
                <param name="context" value="public"/>
                <param name="proxy" value="IP ADDRESS OR DOMAIN"/>
                <param name="register" value="true"/>
                <param name="register-transport" value="tcp (or udp)"/>
                <param name="retry-seconds" value="15"/>
        </gateway>
  </gateways>

This should work... If you get Not Found, try looking at your server's log.

you also should create an outgoing gateway, even if you're not registering,
like:

assuming the outgoing gateway will not auth-challenge you:

        <gateway name="outgoing_gw">
                <param name="username" value="none"/>
                <param name="realm" value="whatever"/>
                <param name="password" value="none"/>
                <param name="context" value="public"/>
                <param name="proxy" value="IP ADDRESS OR DOMAIN"/>
                <param name="register" value="false"/>
                <param name="register-transport" value="tcp (or udp)"/>
                <param name="retry-seconds" value="15"/>
        </gateway>

Adding it to the external is a good idea, so the final config will be:


  <gateways>

        <gateway name="whatever">
                <param name="username" value="1001"/>
                <param name="realm" value="THE REALM AGAINST UPON WHICH THE
HASH WILL BE BUILT"/>
                <param name="password" value="password"/>
                <param name="extension" value="auto_to_user"/>
                <param name="context" value="public"/>
                <param name="proxy" value="IP ADDRESS OR DOMAIN"/>
                <param name="register" value="true"/>
                <param name="register-transport" value="tcp (or udp)"/>
                <param name="retry-seconds" value="15"/>
        </gateway>

        <gateway name="outgoing_gw">
                <param name="username" value="does_not_matter_will_not_be
used_but_must_be_set"/>
                <param name="realm" value="SAME AS PROXY"/>
                <param name="password" value="does_not_matter_will_not_be
used_but_must_be_set"/>
                <param name="context" value="public"/>
                <param name="proxy" value="IP ADDRESS OR DOMAIN"/>
                <param name="register" value="false"/>
                <param name="register-transport" value="tcp (or udp)"/>
                <param name="retry-seconds" value="15"/>
        </gateway>

  </gateways>


Once this is working and you see it REGED, calls will be coming in (as per
"context") on the public dialplan... the you just need to bridge it like:

        <extension name="forward call">
        <condition field="destination_number" expression="^(whatever your
regexp is)$">
        <action application="info"/>
        <action application="bridge" data="sofia/gateway/outgoing_gw/$1"/>
        <action application="hangup"/>
        </condition>
        </extension>

I hope I was able to help..

Enjoy freeSWITCH, it's fantastic!

David

On Thu, Sep 29, 2016 at 9:40 AM, Scholz, Dieter <rd-disc at gmx.net> wrote:

> Hello,
>
> thanks for your comment. The URL you mentioned will be helpful, when I
> reach the dialplan stage.
>
> But at the moment I have problems connecting to the HIPATH maschine. I
> gave some more details in another post.
>
> Dieter
>
>
> Am 28.09.2016 um 17:25 schrieb Stanislav Sinyagin:
> > you need to create one more gateway that will communicate to your SIP
> > provider. This gateway can be configured with username and password if
> > the ITSP requires authentication.
> >
> > Then, you will need to create the dialplan that accepts calls from
> > each of these gateways and routes them to the other gateway.
> >
> > It's quite trivial once you figure out how SIP profiles and dialplan
> > contexts work together.
> >
> > Hopefully this will provide you some hints:
> > https://freeswitch.org/confluence/display/FREESWITCH/
> FreeSWITCH+PBX+Example
> >
> >
> >
> >
> > On Wed, Sep 28, 2016 at 2:22 PM, Scholz, Dieter <rd-disc at gmx.net> wrote:
> >> Hello,
> >>
> >> this is my problem:
> >>
> >> I would like to use Freeswitch as a gateway between our Hipath PBX and
> >> an external SIP provider.
> >>
> >> For that I created a Hipath SIP account that is working (tested with
> >> X-lite). Now I would like to let Freeswitch act as a client using this
> >> SIP account. Whenever there is an incoming call Freeswitch should act on
> >> it and forward this call to an external SIP provider.
> >>
> >> Is this possible? Does that make sense? Are there alternatives?
> >>
> >> At the moment I'm stuck to configure Freeswitch as a client. I added the
> >> Hipath SIP account as a gateway. When I compare the SIP traffic of
> >> X-lite and Freeswitch using Wireshark I found out that Freeswitch dows
> >> not offer the 'Subscribe' option. Is this the reason why it fails?
> >>
> >> Can you help me?
> >>
> >> Thanks in advance.
> >>
> >> Dieter
> >>
> >> ____________________________________________________________
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> >>
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> _____________
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> > http://www.freeswitchsolutions.com
> >
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> >
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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