[Freeswitch-users] Hangup problem/SIP BYE lacking authentication

Wellie Chao wchao at yahoo.com
Fri Oct 31 07:42:27 PDT 2008

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" 
         <action application="set" data="sip_use_gateway=broadview"/>
         <action application="transfer" data="$1 XML default"/>

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" 
       <action application="set" data="sip_use_gateway=broadview"/>
       <action application="transfer" data="1001"/>

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 <anthony.minessale at gmail.com>
> Reply-To: freeswitch-users at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> 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 (  if there was a gateway with either of those names,
> it would find it.
> So try naming your gateway SipLocal or
> 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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