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@gmail.com> wrote:
Hi,
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 192.168.0.1:5060 and my handset callerid is +13035555555 I would like to set the contact to sip:+13035555555@192.168.0.1:5060
Is there an ability within Freeswitch to set the Contact header based on callerid?
Craig
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