[Freeswitch-users] Freeswitch as a SIP client

Scholz, Dieter rd-disc at gmx.net
Thu Sep 29 11:36:52 MSD 2016


Hello,

thanks for your comment.

Perhaps I was a little unspecific. So here are some more details:

I'm a beginner concerning VOIP (I reached a state where I am able to 
connect VOIP endpoints, develop a dialplan and connect to an external 
provider) but a very experienced network and Linux user. Because I would 
like to learn Freeswitch from scratch I started with the minimal 
configuration of the Jessie distribution.

I added a gateway definition to the external profile (tried the internal 
profile, too - same behaviour):

<include>
         <gateway name="x">
                 <param name="realm" value="IP ADDRESS"/>
                 <param name="username" value="USERID" />
                 <param name="password" value="PASSWORD" />
                 <param name="register" value="true" />
         </gateway>
</include>

When I start Freeswitch a connaction attempt is made. A REGISTER is send 
to the HIPATH, a TRYING and then I receive an answer 404 NOTFOUND. If I 
use X-lite and connect to the HIPATH the connection attempt is 
successful. So the SIP account of the HIPATH seems to work.

My sofia.conf.xml is:

<configuration name="sofia.conf" description="sofia endpoint">
         <global_settings>
                 <param name="log-level" value="0"/>
                 <param name="tracelevel" value="DEBUG"/>
         </global_settings>
         <profiles>
                 <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml"/>
         </profiles>
</configuration>

And my external.xml is:

<profile name="external">
         <gateways>
                 <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
         </gateways>
         <settings>
                 <param name="sip-port" value="$${external_sip_port}"/>
                 <param name="auth-calls" value="false"/>
                 <param name="debug" value="0"/>
                 <param name="dialplan" value="XML"/>
                 <param name="context" value="public"/>
                 <param name="codec-prefs" value="$${global_codec_prefs}"/>
                 <param name="rtp-ip" value="$${local_ip_v4}"/>
                 <param name="sip-ip" value="$${local_ip_v4}"/>
                 <param name="ext-rtp-ip" value="auto-nat"/>
                 <param name="ext-sip-ip" value="auto-nat"/>
         </settings>
</profile>

The console says: Failed registration.

Any hints on what is missing? I looked up the vanilla configuration but 
there seems to be no promising additional options.

Thanks.

Dieter


Am 28.09.2016 um 17:24 schrieb Peter Steinbach:
> Hello Dieter,
>
> sure this should work. Registering a Freeswitch as a SIP client to an
> external SIP server can be done through a gateway definition, as you
> pointed out. The first step is to see, if the gateway is registered to
> the Hipath ("sofia status" should show REGED for the gateway). If yes,
> you're almost done. (I assume, you have to register to the Hipath, right?)
> I think, that Freeswitch does not offer Subscribes in the external
> profile, only in the default internal profile.
> Do you receive an INVITE from the Hipath? Also note, that Freeswitch's
> external profile is on Port 5080 instead of 5060, so Hipath has to send
> to 5080.
> On Linux, you may use ngrep to grep the SIP traffic: "ngrep -d any port
> 5080 -W byline").
>
> Best regards
> Peter
>
> On 09/28/16 14:22, Scholz, Dieter 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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