There are two ways. 1 - define a gateway, the contact user will be taken from the gateway parameters. 2 - Set the channel variable "sip_contact_user" in your dialplan, the IP address part of the URL is still controlled by FS, but this will replace the mod_sofia username that appears today. This only works if you do not use a gateway in the sip profile Simon On 8/13/08 12:31 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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