[Freeswitch-users] Call Forwarding Problem

Michael Collins msc at freeswitch.org
Fri Sep 24 15:03:04 PDT 2010


All of our redirect handling information is covered here:

If I were you I would play around with the sip-related variables like
sip_to_uri and sip_to_user when handling the 302 redirect from the Polycom.
Let us know what you find out.


On Wed, Sep 22, 2010 at 5:59 AM, Mike <motosota at gmail.com> wrote:

> Hi,
> Here's a problem I posted about on the IRC some time ago - I said I'd
> do some traces and put them in pastebin, which I've done here:
> http://pastebin.freeswitch.org/13970
> A brief description of the problem and the calls involved in the pastebin.
> Incoming calls are delivered to the Freeswitch from a SIP trunking
> provider who are also used to route outgoing calls. This is not via a
> gateway.
> The SIP trunking provider insists that for outgoing calls, the data in
> the "From" field in the INVITE message contains a PSTN number that
> they have 'associated' with the SIP trunking service,
> e.g.
> From: "Zoltan" <sip:02031375892 at<sip%3A02031375892 at>>
> is OK.
> From: "Zoltan" <sip:anonymous at<sip%3Aanonymous at>>
> would be rejected by
> them, as would
> From: "Zoltan" <sip:01344299800 at<sip%3A01344299800 at>>
> as 01344299800 is
> 'someone elses' number, not one of theirs that they have allocated to
> us.
> The first call in the pastebin shows a Polycom phone making an
> outgoing call that works / is allowed as the
> effective_caller_id_number is set to an "allowed" number 02031375892.
> The second call, starting at line 871 shows a call originating in the
> PSTN from number 01344299800 to the Polycom. The SIP provider delivers
> the call to the FreeSwitch which in turn sends the call to the
> Polycom. The Polycom has been set to divert all incoming calls to
> number 07770380800 and replies with a "SIP/2.0 302 Moved Temporarily"
> with the forwarded to number as the contact. This in turn causes the
> FreeSwitch to generate an INVITE message which is sent to the trunk
> provider with:
> From: "01344299800" <sip:01344299800 at<sip%3A01344299800 at>>
> and
> To: <sip:07770380800 at <sip%3A07770380800 at>>
> Since 01344299800 is not an "allowed" number according to the
> provider, the call is rejected.
> Putting aside the argument for now of whether the SIP provider should
> be carrying out this rather brutal form of "CLI screening" (the data
> in the 'From' field appears as the A Party Number  / CLI once the call
> hits the PSTN / mobile networks), is there any way to modify the
> behaviour of this call so that in response to the "SIP/2.0 302 Moved
> Temporarily" message, the INVITE which is sent out has the number from
> the effective_caller_id_number variable for the forwarding extension,
> rather than the number from the "From" field of the incoming invite?
> Hope this makes sense,
> Mike
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