[Freeswitch-users] problem making internal calls

[Craig Kowald]

Well, I have no idea why, but when I booted this morning it all just
worked without me changing anything :/

Sadly I did not get a log of it, but I did notice that the first time I
ran freeswitch that the IP phone registered. Further, on subsequent
executions of freeswitch, the IP phone does not seem to need/bother to
re-register but I can make internal calls from the analog phone
connected to TDM to the IP phone.

Now I am off to try and sort out calling from the IP phone to the analog
phone. I am getting some errors but have not looked into it yet.

[ERR] mod_openzap.c:719 channel_outgoing_channel() Missing span
[ERR] switch_ivr_originate.c:533 switch_ivr_originate() Cannot Create
Outgoing Channel! cause: DESTINATION_OUT_OF_ORDER                


To answer some questions:

>  Is the phone actually registering to freeswitch?  It appears not.

I have noticed it register once, but not again. Is there something I
should be setting/doing to force it to register every time freeswitch
starts up ?

> in the directory, is the id 5 in the domain 192.1680.0.4?

Yes, id 5 is in the correct domain. In directory.xml I have:
<domain name="$${sip_profile}">


> [change profile from ip address to domain name]

I did not do this - since it all just worked this morning. I will keep
this in mind though if I have further problems.



Thanks for the help.


Regards,
Craig.


On Wed, 2007-08-15 at 11:46 -0400, Jason Garland wrote:

> The SIP profile name is an ip address?
> 
> How about changing it to "foo.localdomain"
> 
> On 8/15/07, Michael Jerris <mike at jerris.com> wrote:
> >
> >  Is the phone actually registering to freeswitch?  It appears not.  Also, in
> > the directory, is the id 5 in the domain 192.1680.0.4?  You will want to
> > look in the log before the call is made to see what happens with the
> > register attempt (or lack of) from the phone.
> >
> >  Mike
> >
> >
> >
> >  On 8/15/07 3:40 AM, "Craig Kowald"
> > <craig.kowald at cubemicrosystems.com> wrote:
> >
> >
> > I am trying to call from one internal phone to another internal phone - but
> > I can not get it working :(.
> >
> >  I have an analog phone connected to a Digium TDM400P and an IP phone
> > (identified as user:5)
> >
> >  In directory.xml I have identified the IP phone as:
> >  <user id="5">
> >   <params>
> >    <param name="password" value="5"/>
> >   </params>
> >  </user>
> >
> >
> >  In my dialplan I have:
> >  <extension name="Internal-5">
> >   <condition field="destination_number" expression="^5$">
> >    <action application="bridge"
> > data="sofia/${sip_profile}/5%${sip_profile}"/>
> >   </condition>
> >  </extension>
> >
> >
> >  In sophia.xml I try to make it so that sip phones have to register by
> > having:
> >  <!--<param name="accept-blind-reg" value="true"/> -->
> >  <param name="auth-calls" value="true"/>
> >  <param name="auth-all-packets" value="true"/>
> >
> >
> >
> >  When I dial '5' on the analog phone I can see that freeswitch tries to make
> > the connection, but can not:
> >
> >  [NOTICE] switch_core_state_machine.c:134
> > switch_core_standard_on_execute() Execute
> > bridge(sofia/${sip_profile}/5%${sip_profile})
> >  [DEBUG] switch_core_state_machine.c:158
> > switch_core_standard_on_execute() Expanded String
> > bridge(sofia/192.168.0.4/5%192.168.0.4)
> >  [WARNING] mod_sofia.c:1397 sofia_outgoing_channel() Cannot locate
> > registered user 5 at 192.168.0.4
> >
> >
> >  Any ideas/suggestions on how to call internal sip phones ?
> >
> >
> >  Regards,
> >  Craig.
> >  ________________________________
> > _______________________________________________
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> >
> >
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