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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Yes, I agree that the called party should not be forcing the
contact, however the second half of your sentence reinforces that I _<i>should</i>_
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).<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Craig<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Anthony
Minessale<br>
<b>Sent:</b> Wednesday, 13 August 2008 11:02 PM<br>
<b>To:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Subject:</b> Re: [Freeswitch-users] setting contact header<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'>That is not currently possible
only because there was no reason to make that configurable and it's not
acceptable.<br>
<br>
The called party should *NOT* be forcing the calling party to use a certain
contact, the contact is the calling parties business.<br>
<br>
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. <br>
<br>
<br>
<br>
<br>
<br>
<o:p></o:p></p>
<div>
<p class=MsoNormal>On Wed, Aug 13, 2008 at 2:31 AM, Craig Guy <<a
href="mailto:craig.a.guy@gmail.com">craig.a.guy@gmail.com</a>> wrote:<o:p></o:p></p>
<div>
<div>
<p>Hi,<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>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 <span style='font-family:Wingdings'>J</span><o:p></o:p></p>
<p> <o:p></o:p></p>
<p>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:<o:p></o:p></p>
<p> <o:p></o:p></p>
<p style='margin-left:54.0pt;text-indent:-18.0pt'>a)
e.164 format<o:p></o:p></p>
<p style='margin-left:54.0pt;text-indent:-18.0pt'>b)
The From, RPID and Contact headers to match<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>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><o:p></o:p></p>
<p> <o:p></o:p></p>
<p>For example if my sip profile is bound to <a href="http://192.168.0.1:5060"
target="_blank">192.168.0.1:5060</a> and my handset callerid is +13035555555 I
would like to set the contact to sip:+13035555555@192.168.0.1:5060<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Is there an ability within Freeswitch to set the Contact header based on
callerid?<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Craig<o:p></o:p></p>
<p> <o:p></o:p></p>
</div>
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<p class=MsoNormal style='margin-bottom:12.0pt'><br>
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<p class=MsoNormal><br>
<br clear=all>
<br>
-- <br>
Anthony Minessale II<br>
<br>
FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>
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<br>
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<br>
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