I tried the following in conf/dialplan/extensions/7_inbound.xml:

  <extension name="broadview_inbound_9325">
    <condition field="destination_number" expression="^12675379325|2675379325$">
      <action application="export" data="sip_use_gateway=broadview"/>
      <action application="transfer" data="1001"/>
    </condition>
  </extension>

Also tried the following in conf/dialplan/public.xml:

    <extension name="public_did_broadview">
      <condition field="destination_number" 
expression="^(12675379324|2675379324|12675379325|2675379325)$">
        <action application="export" data="sip_use_gateway=broadview"/>
        <action application="transfer" data="$1 XML default"/>
      </condition>
    </extension>

Neither helped. When you say add it to the dial string directly that calls it, I'm not sure what you mean (I know the general format of {var_name=var_value}, so that's not my question). Do you mean add it in front of the 1001 as the target of the transfer?

By the way, hangup DOES work properly if I create another gateway and name it 64.115.128.6. However, I'd love to get it working without having to create a duplicate gateway with a non-intuitive name. It's definitely a lot better than nothing to do it that way, but I'd prefer to have it work with the sip_use_gateway scheme you mention. I'm assuming I'm just doing something wrong with how sip_use_gateway should be specified in the XML configuration files. Can you tell what I am doing wrong?

On Fri, 31 Oct 2008, Anthony Minessale wrote:

Date: Fri, 31 Oct 2008 09:49:18 -0500
From: Anthony Minessale <[EMAIL PROTECTED]>
Reply-To: [email protected]
To: [email protected]
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

try using "export" instead of "set" or add it to the dial string directly that 
calls it

{sip_use_gateway=broadview}sofia/.......


On Fri, Oct 31, 2008 at 9:42 AM, Wellie Chao <[EMAIL PROTECTED]> wrote:
      Where do you recommend I put the sip_use_gateway=broadview action?

      I have tried in the conf/dialplan/public.xml like so:

         <extension name="public_did_broadview">
           <condition field="destination_number" 
expression="^(12675379324|2675379324|12675379325|2675379325)$">
             <action application="set" data="sip_use_gateway=broadview"/>
             <action application="transfer" data="$1 XML default"/>
           </condition>
         </extension>

      I've also tried in conf/dialplan/extensions/7_inbound.xml (a file I 
created that is pulled in via an include
      pre-processor directive):

       <extension name="broadview_inbound_9325">
         <condition field="destination_number" 
expression="^12675379325|2675379325$">
           <action application="set" data="sip_use_gateway=broadview"/>
           <action application="transfer" data="1001"/>
         </condition>
       </extension>

      I have a gateway named broadview in conf/sip_profiles/external. In both 
cases, I still get the following error on
      the Freeswitch console:

      2008-10-31 10:37:28 [ERR] sofia_reg.c:1089 
sofia_reg_handle_sip_r_challenge() No Matching gateway found

      On Fri, 31 Oct 2008, Anthony Minessale wrote:

            Date: Fri, 31 Oct 2008 08:04:23 -0500
            From: Anthony Minessale <[EMAIL PROTECTED]>
Reply-To: [email protected]
To: [email protected]
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication

See what they said in the challenge?

WWW-Authenticate: Digest 
realm="SipLocal",nonce="3e952db60fb8",stale=false,algorithm=MD5,qop="auth"

Since this is a spontaneous challenge (which i think is somewhat silly since it 
lets you talk on the phone for 40
minutes then makes you authenticate to hangup but *shrug*) FS does not know 
which gateway to use for credentials.

The realm they sent was SipLocal so FS is looking in its configuration for a 
gateway with that name.
The 2nd thing it tries is the host from the To: header (64.115.128.6).  if 
there was a gateway with either of those
names,
it would find it.

So try naming your gateway SipLocal or 64.115.128.6
or you can try setting the variable sip_use_gateway=<whatever> on the channel 
which can give it a hint which
gateway to use.


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