[Freeswitch-users] setting contact header

Craig Guy craig.a.guy at gmail.com
Wed Aug 13 10:22:09 PDT 2008

Yes, I agree that the called party should not be forcing the contact,
however the second half of your sentence reinforces that I _should_ be able
to change it to whatever I like.  The scenario is that Freeswitch is acting
as an SBC between my UA and a PSTN termination gateway, my UA treats
Freeswitch is an outbound proxy.  The reason that I am using a B2BUA rather
than a proxy is that I need to load balance and failover between 3 gateways,
my UA has no knowledge of the gateways and it is up to Freeswitch to select
a gateway for call completion (and thus rewrite both the request URI and To
header - proxies are not supposed to rewrite the To header as this breaks up
the SIP dialog into multiple call legs, which is a B2BUA function).  In this
instance the termination provider is Level3 and it seems entirely reasonable
that I populate the contact header with a valid e.164 style URI.  This is a
specific case in that I am dealing with a PSTN gateway and the from, rpid
and contact headers are used by Level3 for ANI for purposes of toll free and
billing (eg intra vs inter state calling).




From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Sent: Wednesday, 13 August 2008 11:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] setting contact header


That is not currently possible only because there was no reason to make that
configurable and it's not acceptable.

The called party should *NOT* be forcing the calling party to use a certain
contact, the contact is the calling parties business.

If we did support that it would have to fall under the NDLB "No Device Left
Behind" series of configuration options designed for interop with
less-than-compliant SIP UA and it would be quite hard to do right because in
reality the profile is the true SIP UA not each call.  So the contact
address is the url of the profile that should not change at all among
different calls.  Support for this could cause a regression that breaks 100
other complicated SIP scenarios. 

On Wed, Aug 13, 2008 at 2:31 AM, Craig Guy <craig.a.guy at gmail.com> wrote:



First, have been lurking for about a week and I feel Freeswitch is a great
project, coming from a non-programmer background and heavy familiarity with
Asterisk and Callweaver it's a bit of a learning curve to come to grips with
the XML but I'm slowly getting there J


I'm currently doing some interop testing with Freeswitch and the provider
has come back to me stating that they require the Contact header to exhibit:


a)      e.164 format

b)      The From, RPID and Contact headers to match


Try as I might I am seemingly unable to set the user portion of the Contact
header for the outbound leg to the termination provider.  The contact is
always mod_sofia@<ip:port>.  I have found the sip-force-contact setting
however it seems to have no effect on the contact header (and I might be
using it incorrectly in any case).  The ideal behaviour for me is for the
contact to take the form of callerid@<ip:port>


For example if my sip profile is bound to and my handset
callerid is +13035555555 I would like to set the contact to
sip:+13035555555 at


Is there an ability within Freeswitch to set the Contact header based on




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Anthony Minessale II

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