[Freeswitch-users] Intercom

[andrew]

That didn't work. For some reason as soon as I added
the ";intercom=true" to the end of the line it tries
to call 81000 not 1000 and the call fails. Here is
what I have after your change to sofia from user. This
works fine without the ;intercom=true.


   <extension name="extension-intercom">
      <condition field="destination_number"
expression="^8(10[01][0-9])$">
	<action application="set" data="dialed_ext=$1"/>
	<action
application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action>
	<action application="export"
data="sip_auto_answer=true"/>
	<action application="bridge"
data="sofia/$${domain}/${dialed_ext}"/>
      </condition>
    </extension>
	
	
I need the sip packet to look like this
	
	
	
U 192.168.1.10:5060 -> 192.168.1.102:5060
INVITE sip:204 at 192.168.1.102:5060;intercom=true
SIP/2.0.
Via: SIP/2.0/UDP
192.168.1.10:5060;branch=z9hG4bK65cb9af7;rport.
From: "Office" <sip:200 at 192.168.1.10>;tag=as00afe79d.
To: <sip:204 at 192.168.1.102:5060;intercom=true>.
Contact: <sip:200 at 192.168.1.10>.
Call-ID:
7190aec117c498d615b777fc129ab479 at 192.168.1.10.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Fri, 07 Mar 2008 01:40:35 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY.
Supported: replaces.
Call-Info: <uri>;answer-after=0.
Alert-Info: Ring Answer.
Content-Type: application/sdp.
Content-Length: 262.




With the ;intercom=true in two places the "Call-Info:
<uri>;answer-after=0" and the "Alert-Info: Ring
Answer" Is there a way to make this work?




--- Brian West <brian.west at mac.com> wrote:

> 
> >
> >
> >
>
-------------------------------------------------------
> >   <extension name="extension-intercom">
> >      <!-- <condition field="${sip_to_params}"
> > expression="intercom\=true"/> -->
> >      <condition field="destination_number"
> > expression="^8(10[01][0-9])$">
> > 	<action application="set" data="dialed_ext=$1"/>
> > 	<action
> >
>
application="export"><![CDATA[sip_h_Call-Info=<sip:$$
> 
> > {domain}>;answer-after=0]]></action>
> > 	<action application="export"
> > data="sip_auto_answer=true"/>
> > 	<action application="bridge"
> >
> data="user/${dialed_ext}@$${domain};intercom=true"/>
> >      </condition>
> >    </extension>
> >
>
-------------------------------------------------------
> >
> > and now i just get a busy signal when I call. Any
> > idea?
> >
> >
> 
> 
> URI params aren't passed on the user channel
> 
> data="sofia/$${domain}/${dialed_ext};intercom=true
> 
> Then you'll need to add
> 
> <action application="export"
> data="sip_h_Alert-Info=Ring Answer"/>
> 
> /b
> 
> 
> >
> > --- Brian West <brian.west at mac.com> wrote:
> >
> >> I think if you call a phone ie
> >> sofia/domain/user;intercom=true
> >>
> >> The param will get passed also....
> >>
> >> /b
> >>
> >> On Mar 6, 2008, at 1:51 PM, andrew wrote:
> >>
> >>> intercom=true
> >>
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
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> >>
> >
>
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> >>
> >
>
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> >>
> >
> >
> >
> >       
> >
>
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