static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, char *vxml_url)
{
        char invite[256]="";
        char from[256];
        char to[256];
        char tmp[80];
        char iabuf[INET_ADDRSTRLEN];
        char *l = default_callerid, *n=NULL;
        int x;
        char urioptions[256]="";

        if (ast_test_flag(p, SIP_USEREQPHONE)) {
                char onlydigits = 1;
                x=0;

                /* Test p->username against allowed characters in AST_DIGIT_ANY
                If it matches the allowed characters list, then sipuser = ";user=phone"
                If not, then sipuser = ""
                */
                /* + is allowed in first position in a tel: uri */
                if (p->username && p->username[0] == '+')
                        x=1;

                for (; x<strlen(p->username); x++) {
                        if (!strchr(AST_DIGIT_ANYNUM, p->username[x])) {
                                onlydigits = 0;
                                break;
                        }
                }

                /* If we have only digits, add ;user=phone to the uri */
                if (onlydigits)
                        strcpy(urioptions, ";user=phone");
        }


        snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);

        if (p->owner) {
                l = p->owner->cid.cid_num;
                n = p->owner->cid.cid_name;
        }
        if (!l || (!ast_isphonenumber(l) && default_callerid[0]))
                        l = default_callerid;
        /* if user want's his callerid restricted */
        if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
                l = CALLERID_UNKNOWN;
                n = l;
        }
        if (!n || ast_strlen_zero(n))
                n = l;
        /* Allow user to be overridden */
        if (!ast_strlen_zero(p->fromuser))
                l = p->fromuser;
        else /* Save for any further attempts */
                ast_copy_string(p->fromuser, l, sizeof(p->fromuser));

        /* Allow user to be overridden */
        if (!ast_strlen_zero(p->fromname))
                n = p->fromname;
        else /* Save for any further attempts */
                ast_copy_string(p->fromname, n, sizeof(p->fromname));

        if ((ourport != 5060) && ast_strlen_zero(p->fromdomain))        /* Needs to be 5060 */
                snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, ourport, p->tag);
        else
                snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, p->tag);

        /* If we're calling a registred SIP peer, use the fullcontact to dial to the peer */
        if (!ast_strlen_zero(p->fullcontact)) {
                /* If we have full contact, trust it */
                char *scan, *addr, *port_str, *expiry_str, *username, *contact;
                /* try to split this fullcontact into the invite contact */
                scan=p->fullcontact; /*ast_copy_string(scan,p->fullcontact,sizeof(scan));*/
                addr = strsep(&scan, ":");
                port_str = strsep(&scan, ":");
                expiry_str = strsep(&scan, ":");
                username = strsep(&scan, ":");
                contact = scan; /* Contact include sip: and has to be the last part of the database entry as long as we use : as a separator */
                ast_log(LOG_DEBUG,"contact is: %s\n",contact);

                ast_copy_string(invite,contact,sizeof(contact));
                ast_log(LOG_DEBUG,"Our invite is still:%s",invite);

                /*ast_copy_string(invite, p->fullcontact, sizeof(invite));*/
        /* Otherwise, use the username while waiting for registration */
        } else if (!ast_strlen_zero(p->username)) {
                if (ntohs(p->sa.sin_port) != 5060) {            /* Needs to be 5060 */
                        snprintf(invite, sizeof(invite), "sip:%s@%s:%d%s",p->username, p->tohost, ntohs(p->sa.sin_port), urioptions);
                } else {
                        snprintf(invite, sizeof(invite), "sip:%s@%s%s",p->username, p->tohost, urioptions);
                }
        } else if (ntohs(p->sa.sin_port) != 5060) {             /* Needs to be 5060 */
                snprintf(invite, sizeof(invite), "sip:%s:%d%s", p->tohost, ntohs(p->sa.sin_port), urioptions);
        } else {
                snprintf(invite, sizeof(invite), "sip:%s%s", p->tohost, urioptions);
        }
        ast_copy_string(p->uri, invite, sizeof(p->uri));
        /* If there is a VXML URL append it to the SIP URL */
        if (vxml_url)
        {
                snprintf(to, sizeof(to), "<%s>;%s", invite, vxml_url);
        } else {
                snprintf(to, sizeof(to), "<%s>", invite);
        }
        memset(req, 0, sizeof(struct sip_request));
        ast_log(LOG_DEBUG,"Our invite is still:%s",invite);
        init_req(req, sipmethod, invite);
        snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);

        add_header(req, "Via", p->via);
        /* SLD: FIXME?: do Route: here too?  I think not cos this is the first request.
         * OTOH, then we won't have anything in p->route anyway */
        add_header(req, "From", from);
        ast_copy_string(p->exten, l, sizeof(p->exten));
        build_contact(p);
        ast_log(LOG_DEBUG,"ADDING NEW HEADER:%s\n",to);
        add_header(req, "To", to);
        add_header(req, "Contact", p->our_contact);
        add_header(req, "Call-ID", p->callid);
        add_header(req, "CSeq", tmp);
        add_header(req, "User-Agent", default_useragent);
}